Re: [asterisk-users] SDP a=ice-ufrag & a=ice-pwd UNSUPPORTED OR FAILED
Hello List I would very much like to have some feedback on this. Where do I have to look ? Is it in the Asterisk version (13.38.3) maybe ? Is it for sure in my config ?! Kind regards. Op 28/06/2023 om 16:14 schreef Jonas Kellens: Hello list when trying to set up webRTC communications with sipjs client package (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file the following : DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 99.88.77.66... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=candidate:4275385644 1 udp 2122260223 192.168.0.18 57987 typ host generation 0 network-id 1... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=candidate:3686672562 1 udp 2122194687 172.21.64.1 57988 typ host generation 0 network-id 2... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=candidate:4292167434 1 udp 1686052607 99.88.77.66 57987 typ srflx raddr 192.168.0.18 rport 57987 generation 0 network-id 1... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=candidate:8380856 1 tcp 1518280447 192.168.0.18 9 typ host tcptype active generation 0 network-id 1... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=candidate:622132262 1 tcp 1518214911 172.21.64.1 9 typ host tcptype active generation 0 network-id 2... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=ice-ufrag:zBkv... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=ice-pwd:8LVdZW/AEq7hp898bLtsI/5W... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=ice-options:trickle... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=fingerprint:sha-256 92:6B:C7:79:41:B1:42:78:2B:3A:75:8B:0B:D0:C7:4C:7C:4E:4F:2D:03:A2:DA:D9:BB:CE:B2:39:5D:20:A0:EF... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=setup:actpass... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=mid:0... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=msid:4f4db37d-65ff-4f57-8c1c-b404f976c3fb cc4a3d72-3e9d-4926-b57c-056b6e7a6d6c... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtcp-mux... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 opus/48000/2... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtcp-fb:111 transport-cc... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=fmtp:111 minptime=10;useinbandfec=1... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:63 red/48000/2... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=fmtp:63 111/111... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:13 CN/8000... OK in sip.conf I have : icesupport = yes in rtp.conf I have : icesupport=true stunaddr=stun.ekiga.net sip peer has everything set for webrtc : Realtime peer: Yes, cached Prim.Transp. : WS Allowed.Trsp : WSS Codecs : (alaw|g729|gsm) Useragent : SIP.js/0.10.0 Reg. Contact : sip:u79mer6v@1u7hp86jdg67.invalid;transport=ws RTP Engine : asterisk Encryption : Yes RTCP Mux : Yes avpf = yes force_avp =yes icesupport = yes dtlsenable = yes dtlsverify = fingerprint dtlssetup = actpass dtlsfingerprint
[asterisk-users] SDP a=ice-ufrag & a=ice-pwd UNSUPPORTED OR FAILED
Hello list when trying to set up webRTC communications with sipjs client package (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file the following : DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 99.88.77.66... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=candidate:4275385644 1 udp 2122260223 192.168.0.18 57987 typ host generation 0 network-id 1... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=candidate:3686672562 1 udp 2122194687 172.21.64.1 57988 typ host generation 0 network-id 2... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=candidate:4292167434 1 udp 1686052607 99.88.77.66 57987 typ srflx raddr 192.168.0.18 rport 57987 generation 0 network-id 1... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=candidate:8380856 1 tcp 1518280447 192.168.0.18 9 typ host tcptype active generation 0 network-id 1... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=candidate:622132262 1 tcp 1518214911 172.21.64.1 9 typ host tcptype active generation 0 network-id 2... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=ice-ufrag:zBkv... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=ice-pwd:8LVdZW/AEq7hp898bLtsI/5W... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=ice-options:trickle... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=fingerprint:sha-256 92:6B:C7:79:41:B1:42:78:2B:3A:75:8B:0B:D0:C7:4C:7C:4E:4F:2D:03:A2:DA:D9:BB:CE:B2:39:5D:20:A0:EF... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=setup:actpass... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=mid:0... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=msid:4f4db37d-65ff-4f57-8c1c-b404f976c3fb cc4a3d72-3e9d-4926-b57c-056b6e7a6d6c... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtcp-mux... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 opus/48000/2... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtcp-fb:111 transport-cc... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=fmtp:111 minptime=10;useinbandfec=1... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:63 red/48000/2... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=fmtp:63 111/111... UNSUPPORTED OR FAILED. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:13 CN/8000... OK in sip.conf I have : icesupport = yes in rtp.conf I have : icesupport=true stunaddr=stun.ekiga.net sip peer has everything set for webrtc : Realtime peer: Yes, cached Prim.Transp. : WS Allowed.Trsp : WSS Codecs : (alaw|g729|gsm) Useragent : SIP.js/0.10.0 Reg. Contact : sip:u79mer6v@1u7hp86jdg67.invalid;transport=ws RTP Engine : asterisk Encryption : Yes RTCP Mux : Yes avpf = yes force_avp =yes icesupport = yes dtlsenable = yes dtlsverify = fingerprint dtlssetup = actpass dtlsfingerprint = sha-256 Why is there "UNSUPPORTED OR FAILED" in the log when processing "a=ice-ufrag" and "ice-pwd" ?? Asterisk gives no "a=ice-ufrag" and "ice-pwd" in its "SIP/2.0 200 OK" response to the INVITE and thus sipjs aborts the SIP call with a 488-
Re: [asterisk-users] Discrepancy between Asterisk console and Asterisk Manager DeviceStateChange
On Fri, Feb 11, 2022 at 9:31 AM Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello I notice a major difference in what Asterisk console is telling me (which seems correct) and what Asterisk Manager is telling. A SIP user is called, and the phone does not ring. This is the situation. On Asterisk console I see (which seems to be in line with an unreachable phone) : [Feb 11 11:31:31] VERBOSE[15653][C-0319] app_dial.c: Called SIP/mysipuser6 [Feb 11 11:31:37] VERBOSE[15653][C-0319] app_dial.c: Everyone is busy/congested at this time (1:0/0/1) [Feb 11 11:31:37] VERBOSE[15653][C-0319] pbx.c: Executing [202@from-PBX:253] NoOp("SIP/mysipuser12-157d", "DIALSTATUS=CHANUNAVAIL") in new stack However on Asterisk Manager interface I see the event : 11:31:31 Array ( [0] => Event: DeviceStateChange [1] => Privilege: call,all [2] => SystemName: voipserver1 [3] => Device: SIP/mysipuser6 [4] => State: RINGING ) I can reproduce this easily every time : [Feb 11 11:31:46] VERBOSE[15719][C-031a] app_dial.c: Called SIP/mysipuser6 [Feb 11 11:31:53] VERBOSE[15719][C-031a] app_dial.c: Everyone is busy/congested at this time (1:0/0/1) [Feb 11 11:31:53] VERBOSE[15719][C-031a] pbx.c: Executing [202@from-PBX:253] NoOp("SIP/mysipuser12-157f", "DIALSTATUS=CHANUNAVAIL") in new stack 11:31:46 Array ( [0] => Event: DeviceStateChange [1] => Privilege: call,all [2] => SystemName: voipserver1 [3] => Device: SIP/mysipuser6 [4] => State: RINGING ) Why is Asterisk Manager reporting a RINGING state if there is no SIP 180 RINGING received ?! When issuing a SIP DEBUG, I see a SIP INVITE but no response (so no SIP 180 or 183). The answer seems to be, because that's the way chan_sip was written. As soon as an outgoing call is attempted it sets some internal state to ringing, which is then used when it reports device state information. DeviceStateChange is just reporting what chan_sip told it. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com <http://www.sangoma.com> and www.asterisk.org <http://www.asterisk.org> So if "DeviceStateChange" is not reporting the real state of a SIP user/device (like 180-ringing), which event does ?! Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Discrepancy between Asterisk console and Asterisk Manager DeviceStateChange
Hello I notice a major difference in what Asterisk console is telling me (which seems correct) and what Asterisk Manager is telling. A SIP user is called, and the phone does not ring. This is the situation. On Asterisk console I see (which seems to be in line with an unreachable phone) : [Feb 11 11:31:31] VERBOSE[15653][C-0319] app_dial.c: Called SIP/mysipuser6 [Feb 11 11:31:37] VERBOSE[15653][C-0319] app_dial.c: Everyone is busy/congested at this time (1:0/0/1) [Feb 11 11:31:37] VERBOSE[15653][C-0319] pbx.c: Executing [202@from-PBX:253] NoOp("SIP/mysipuser12-157d", "DIALSTATUS=CHANUNAVAIL") in new stack However on Asterisk Manager interface I see the event : 11:31:31 Array ( [0] => Event: DeviceStateChange [1] => Privilege: call,all [2] => SystemName: voipserver1 [3] => Device: SIP/mysipuser6 [4] => State: RINGING ) I can reproduce this easily every time : [Feb 11 11:31:46] VERBOSE[15719][C-031a] app_dial.c: Called SIP/mysipuser6 [Feb 11 11:31:53] VERBOSE[15719][C-031a] app_dial.c: Everyone is busy/congested at this time (1:0/0/1) [Feb 11 11:31:53] VERBOSE[15719][C-031a] pbx.c: Executing [202@from-PBX:253] NoOp("SIP/mysipuser12-157f", "DIALSTATUS=CHANUNAVAIL") in new stack 11:31:46 Array ( [0] => Event: DeviceStateChange [1] => Privilege: call,all [2] => SystemName: voipserver1 [3] => Device: SIP/mysipuser6 [4] => State: RINGING ) Why is Asterisk Manager reporting a RINGING state if there is no SIP 180 RINGING received ?! When issuing a SIP DEBUG, I see a SIP INVITE but no response (so no SIP 180 or 183). Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired
Hello Joshua could it be a bug ? I am using asterisk-certified-13.21-cert6 Kind regards. J. Op 01-07-21 om 20:20 schreef Joshua C. Colp: On Thu, Jul 1, 2021 at 3:15 PM Jonas Kellens <mailto:jonas.kell...@telenet.be>> wrote: Hello Joshua this is the SIP REGISTER at 11:10:45 REGISTER sip:tstv7.domain.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bKcfd4c09d669ce595351ea3b2aba3e245;rport From: <mailto:sip:testacc7700...@tstv7.domain.tld>;tag=3630891428 To: <mailto:sip:testacc7700...@tstv7.domain.tld> Call-ID: 3270725701@192_168_1_18 CSeq: 452 REGISTER Contact: <mailto:sip:testacc7700105@192.168.1.18:5060> Authorization: Digest username="testacc7700105", realm="tstv7.domain.tld", algorithm=MD5, uri="sip:tstv7.domain.tld", nonce="42a70292", response="e2945dacd2d95b47a4801b2471070702" Max-Forwards: 70 User-Agent: C610 IP/42.075.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 I see here "180". I also see in the SIP debug traffic that a SIP REGISTER occurs every 180 seconds, which is what is set in the SIP client. This "Expired" notice occured only once at 11:20:55. How come this happens only once ? And why should there *ever* be an "Expired" if there is a SIP REGISTER every 180 seconds ?! If what you are saying is correct, then I do not know. The chan_sip module decided that it should be expired. Why that is, I do not know. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com <http://www.sangoma.com> and www.asterisk.org <http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired
Hello Joshua this is the SIP REGISTER at 11:10:45 REGISTER sip:tstv7.domain.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bKcfd4c09d669ce595351ea3b2aba3e245;rport From: ;tag=3630891428 To: Call-ID: 3270725701@192_168_1_18 CSeq: 452 REGISTER Contact: Authorization: Digest username="testacc7700105", realm="tstv7.domain.tld", algorithm=MD5, uri="sip:tstv7.domain.tld", nonce="42a70292", response="e2945dacd2d95b47a4801b2471070702" Max-Forwards: 70 User-Agent: C610 IP/42.075.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 I see here "180". I also see in the SIP debug traffic that a SIP REGISTER occurs every 180 seconds, which is what is set in the SIP client. This "Expired" notice occured only once at 11:20:55. How come this happens only once ? And why should there *ever* be an "Expired" if there is a SIP REGISTER every 180 seconds ?! Kind regards. Op 01-07-21 om 17:41 schreef Joshua C. Colp: On Thu, Jul 1, 2021 at 12:34 PM Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello Joshua these are the 2 previous events on the Manager interface : 2021-06-30 11:10:45 Array ( [0] => Event: PeerStatus [1] => Privilege: system,all [2] => SystemName: tstv7 [3] => ChannelType: SIP [4] => Peer: SIP/testacc7700921 [5] => PeerStatus: Registered [6] => Address: my.lo.cal.ip:55014 ) 2021-06-30 11:10:48 Array ( [0] => Event: PeerStatus [1] => Privilege: system,all [2] => SystemName: tstv7 [3] => ChannelType: SIP [4] => Peer: SIP/testacc7700921 [5] => PeerStatus: Reachable [6] => ) So there is a re-register at 11:10:45 How do you explain the "Expired" 10 minutes later ?? Without the actual SIP REGISTER traffic to show how long the registration was for, I can't really say anything further. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com <http://www.sangoma.com> and www.asterisk.org <http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired
Hello Joshua these are the 2 previous events on the Manager interface : 2021-06-30 11:10:45 Array ( [0] => Event: PeerStatus [1] => Privilege: system,all [2] => SystemName: tstv7 [3] => ChannelType: SIP [4] => Peer: SIP/testacc7700921 [5] => PeerStatus: Registered [6] => Address: my.lo.cal.ip:55014 ) 2021-06-30 11:10:48 Array ( [0] => Event: PeerStatus [1] => Privilege: system,all [2] => SystemName: tstv7 [3] => ChannelType: SIP [4] => Peer: SIP/testacc7700921 [5] => PeerStatus: Reachable [6] => ) So there is a re-register at 11:10:45 How do you explain the "Expired" 10 minutes later ?? Op 30-06-21 om 20:32 schreef Joshua C. Colp: On Wed, Jun 30, 2021 at 3:28 PM Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello I see the following event from the Asterisk Manager : 2021-06-30 11:20:55 Array ( [0] => Event: PeerStatus [1] => Privilege: system,all [2] => SystemName: tstv7 [3] => ChannelType: SIP [4] => Peer: SIP/testacc7700921 [5] => PeerStatus: Unregistered [6] => Cause: Expired ) The cause is in this message, the registration expired. A re-registration did not occur before the registration expiration so it expired and was unregistered. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com <http://www.sangoma.com> and www.asterisk.org <http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired
Hello I see the following event from the Asterisk Manager : 2021-06-30 11:20:55 Array ( [0] => Event: PeerStatus [1] => Privilege: system,all [2] => SystemName: tstv7 [3] => ChannelType: SIP [4] => Peer: SIP/testacc7700921 [5] => PeerStatus: Unregistered [6] => Cause: Expired ) But I see no SIP REGISTER with SIP-header Expires:0 (so an UNregister if you like) in my SIP debug. What I do see is a SIP OPTION, following a SIP 200 OK (so this is the qualify frequenty) at 11:20:45 [Jun 30 11:20:45] VERBOSE[1581] chan_sip.c: Reliably Transmitting (NAT) to my.lo.cal.ip:55014: OPTIONS sip:testacc7700921@192.168.1.9:5060 SIP/2.0 Via: SIP/2.0/UDP my.aste.risk.ip:5060;branch=z9hG4bK357689ec;rport Max-Forwards: 70 From: "asterisk" ;tag=as0da9cbe1 To: Contact: Call-ID: 0b8c7f5745051ae73193231f5a487...@my.aste.risk.ip:5060 CSeq: 102 OPTIONS User-Agent: TSTv7 Date: Wed, 30 Jun 2021 09:20:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jun 30 11:20:45] VERBOSE[1581] chan_sip.c: <--- SIP read from UDP:my.lo.cal.ip:55014 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP my.aste.risk.ip:5060;branch=z9hG4bK357689ec;rport=5060 From: "asterisk" ;tag=as0da9cbe1 To: ;tag=2924269434 Call-ID: 0b8c7f5745051ae73193231f5a487...@my.aste.risk.ip:5060 CSeq: 102 OPTIONS User-Agent: Yealink SIP-T46G 28.83.0.120 Content-Length: 0 In sip.conf I have the following config concerning SIP registration expiry : maxexpiry=3600 ;minexpiry=60 ;defaultexpiry=120 ;submaxexpiry=3600 ;subminexpiry=60 qualifyfreq=120 So my question is : what causes the Asterisk Manager to report a "PeerStatus: Unregistered" if I find no such data in my SIP debug information ?? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Difference in information realtime database table and Asterisk sip show peers
Hello can anyone explain to me why (and HOW) there is a difference in data between the Asterisk console "sip show peers" and the realtime MySQL configuration ? Using : asterisk-certified-13.21-cert6 Asterisk console data : /usr/sbin/asterisk -rx 'sip show peers' | grep 660091086 660091086/660091086 11.22.33.44 D Yes Yes 55018 OK (31 ms) Cached RT /usr/sbin/asterisk -rx 'sip show peer 660091086' * Name : 660091086 Description : Realtime peer: Yes, cached Secret : DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 11.22.33.44:55018 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 660091086 SIP Options : (none) Codecs : (alaw|g729|gsm) Auto-Framing : No Status : OK (31 ms) Useragent : Cisco/SPA508G-7.5.2 Reg. Contact : sip:660091086@192.168.1.12:5064 Qualify Freq : 12 ms Keepalive : 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No RTCP Mux : Yes MySQL database table data : SELECT name, host, nat, type, qualify, fullcontact, ipaddr, port, regserver, regseconds, lastms, defaultuser FROM sip_buddies WHERE defaultuser='660091086' +---+-+-++-+-++--+---+++-+ | name | host | nat | type | qualify | fullcontact | ipaddr | port | regserver | regseconds | lastms | defaultuser | +---+-+-++-+-++--+---+++-+ | 660091086 | dynamic | force_rport,comedia | friend | yes | | | 0 | | 0 | 0 | 660091086 | +---+-+-++-+-++--+---+++-+ According to Asterisk console, the SIP peer is Registered and qualify (SIP OPTION) is fine (31 ms). According to Mysql data, the SIP peer has no "fullcontact", no "ipaddr", no "port" and 0 (zero) regseconds and 0 (zero) lastms. While I would expect this to be filled in and/or has values higher than 0 (zero). Is there an explanation for this "difference" in data ?! Kind regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi-linux-complete-3.1.0+3.1.0 : issue on CentOS 7.9 but ot on CentOS 6.10
Hello List to answer my own question, and for whom it may interest, I no longer have the error about libtonezone.so with Dahdi version : dahdi-linux-complete-2.11.1+2.11.1 I don't know what the difference is between Dahdi 2.x and Dahdi 3.x but I can say that THERE IS somewhere a difference, as I experienced on Centos 7.9. Kind regards. Op 12-02-21 om 19:11 schreef Jonas Kellens: Hello list when installing latest DAHDI (dahdi-linux-complete-3.1.0+3.1.0) for usage with asterisk-certified-13.21-cert6 on CentOS 6.10 all works well when starting dahdi with "/sbin/service dahdi start". But when installing the same DAHDI version in CentOS 7.9 I get the error :*/usr/sbin/dahdi_cfg: error while loading shared libraries: libtonezone.so.2: cannot open shared object file: No such file or directory* when issuing "systemctl start dahdi.service" Is there something missing on my CentOS 7.9 system to work with the latest DAHDI version ? Or is there a better DAHDI version to be used on CentOS 7.9 ? libtonezone is present on my CentOS 7.9 system : [root@server admin]# locate libtonezone /usr/lib/libtonezone.a /usr/lib/libtonezone.la /usr/lib/libtonezone.so /usr/lib/libtonezone.so.1 /usr/lib/libtonezone.so.1.0 /usr/lib/libtonezone.so.2 /usr/lib/libtonezone.so.2.0 /usr/lib/libtonezone.so.2.0.0 Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi-linux-complete-3.1.0+3.1.0 : issue on CentOS 7.9 but ot on CentOS 6.10
Hello list when installing latest DAHDI (dahdi-linux-complete-3.1.0+3.1.0) for usage with asterisk-certified-13.21-cert6 on CentOS 6.10 all works well when starting dahdi with "/sbin/service dahdi start". But when installing the same DAHDI version in CentOS 7.9 I get the error :*/usr/sbin/dahdi_cfg: error while loading shared libraries: libtonezone.so.2: cannot open shared object file: No such file or directory* when issuing "systemctl start dahdi.service" Is there something missing on my CentOS 7.9 system to work with the latest DAHDI version ? Or is there a better DAHDI version to be used on CentOS 7.9 ? libtonezone is present on my CentOS 7.9 system : [root@server admin]# locate libtonezone /usr/lib/libtonezone.a /usr/lib/libtonezone.la /usr/lib/libtonezone.so /usr/lib/libtonezone.so.1 /usr/lib/libtonezone.so.1.0 /usr/lib/libtonezone.so.2 /usr/lib/libtonezone.so.2.0 /usr/lib/libtonezone.so.2.0.0 Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] defaultexpiry & maxexpiry on peer level
Hello registration time is set to low value because when a network interuption occurs, it takes long time for the endpoint (Phone,...) to re-register. That is my expercience. But about my question : is there a "on peer level" setting possible ? Op 08-10-19 om 19:40 schreef G.Jacobsen: Why do you want such minimal registration time? On Tuesday, 8 October 2019, 17:23:03 EEST, Jonas Kellens wrote: Hello is it possible to determine the SIP.conf parameters 'defaultexpirty' and 'maxexpiry' on a peer basis ? My default value is 300 seconds, but some specific SIP-clients can only send a SIP REGISTER every 3600 seconds. In current configuration these SIP peers now become "Unreachable" after 300 seconds. Or is there another way to differentiate ? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com <http://www.api-digital.com%20>-- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] defaultexpiry & maxexpiry on peer level
Hello is it possible to determine the SIP.conf parameters 'defaultexpirty' and 'maxexpiry' on a peer basis ? My default value is 300 seconds, but some specific SIP-clients can only send a SIP REGISTER every 3600 seconds. In current configuration these SIP peers now become "Unreachable" after 300 seconds. Or is there another way to differentiate ? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk manager : core show hints
Hello I see on the CLI : tst*CLI> core show hints -= Registered Asterisk Dial Plan Hints =- 50@blf : SIP/testacc7 State:Idle Watchers 3 6001@blf : Custom:q-6001 State:Idle Watchers 1 5@blf : SIP/testacc6 State:Unavailable Watchers 1 Is there a way to get this info through the manager API ? Kind regards Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BLF NOTIFY Subscription-State: terminated; reason=timeout
Hello I notice that BLF-buttons on my IP-phone are greyed out and again active after some time. This goes on and on... When looking at Asterisk CLI I see in the SIP NOTIFY : Subscription-State: terminated;reason=timeout The BLF-buttons turn on again after a new SIP SUBSCRIBE from my IP-phone. This SUBSCRIBE happens every 120 seconds. They fade out after about 85 seconds. Then after 35 seconds they come back up. In sip.conf I have the following settings : maxexpiry=100 minexpiry=60 defaultexpiry=100 But that does not seem to change much. Except for SIP REGISTER's : they now happen every 50 seconds (yes, 50) in stead of every 120 seconds. Can anyone help me to find the right timer to tune so I can have stable BLF buttons ? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video
Hello is this mailing list still active ? Op 10-05-19 om 14:10 schreef Jonas Kellens: Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition : Video Support: Yes Prim.Transp. : WS Allowed.Trsp : WSS SIP Options : (none) Codecs : (opus|h264) Status : OK (75 ms) Useragent : SIP.js/0.12.0 Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss RTP Engine : asterisk Encryption : Yes RTCP Mux : Yes Video Support: Yes Prim.Transp. : WS Allowed.Trsp : WSS SIP Options : (none) Codecs : (opus|h264) Status : OK (47 ms) Useragent : SIP.js/0.12.0 Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss RTP Engine : asterisk Encryption : Yes RTCP Mux : Yes In general sip.conf I have : videosupport=yes disallow=all allow=alaw allow=opus allow=h264 When one peer makes a SIP INVITE for a video call, it is clear to me that the necessary codec information is present (this all looks fine to me) : (calling webRTC client) SIP Debugging Enabled for IP: 99.99.255.55 [May 10 10:45:24] [May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 ---> [May 10 10:45:24] INVITE sip:1...@wss.mydomain.tld SIP/2.0 [May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692 [May 10 10:45:24] Max-Forwards: 70 [May 10 10:45:24] To: [May 10 10:45:24] From: "WC User Chrome" ;tag=sdmbqkquhe [May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu [May 10 10:45:24] CSeq: 4132 INVITE [May 10 10:45:24] Contact: [May 10 10:45:24] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [May 10 10:45:24] Supported: outbound [May 10 10:45:24] User-Agent: SIP.js/0.12.0 [May 10 10:45:24] Content-Type: application/sdp [May 10 10:45:24] Content-Length: 5098 [May 10 10:45:24] [May 10 10:45:24] v=0 [May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1 [May 10 10:45:24] s=- [May 10 10:45:24] t=0 0 [May 10 10:45:24] a=group:BUNDLE audio video [May 10 10:45:24] a=msid-semantic: WMS I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E [May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 [May 10 10:45:24] c=IN IP4 99.99.255.55 [May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0 [May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 34197 typ host generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 34197 typ srflx raddr 192.168.1.110 rport 34197 generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192.168.1.110 9 typ host tcptype active generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=ice-ufrag:y8md [May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le [May 10 10:45:24] a=ice-options:trickle [May 10 10:45:24] a=fingerprint:sha-256 C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B [May 10 10:45:24] a=setup:actpass [May 10 10:45:24] a=mid:audio [May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level [May 10 10:45:24] a=sendrecv [May 10 10:45:24] a=rtcp-mux [May 10 10:45:24] a=rtpmap:111 opus/48000/2 [May 10 10:45:24] a=rtcp-fb:111 transport-cc [May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1 [May 10 10:45:24] a=rtpmap:103 ISAC/16000 [May 10 10:45:24] a=rtpmap:104 ISAC/32000 [May 10 10:45:24] a=rtpmap:9 G722/8000 [May 10 10:45:24] a=rtpmap:0 PCMU/8000 [May 10 10:45:24] a=rtpmap:8 PCMA/8000 [May 10 10:45:24] a=rtpmap:106 CN/32000 [May 10 10:45:24] a=rtpmap:105 CN/16000 [May 10 10:45:24] a=rtpmap:13 CN/8000 [May 10 10:45:24] a=rtpmap:110 telephone-event/48000 [May 10 10:45:24] a=rtpmap:112 telephone-event/32000 [May 10 10:45:24] a=rtpmap:113 telephone-event/16000 [May 10 10:45:24] a=rtpmap:126 telephone-event/8000 [May 10 10:45:24] a=ssrc:401971016 cname:cd1IocMPYzY4lNYJ [May 10 10:45:24] a=ssrc:401971016 msid:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E f8eee8bd-dd47-4c14-866d-07069cab255f [May 10 10:45:24] a=ssrc:401971016 mslabel:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E [May 10 10:45:24] a=ssrc:401971016 label:f8eee8bd-dd47-4c14-866d-07069cab255f [May 10 10:45:24] m=video 48086 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 127 122 125 107 108 109 124 [May 10 10:45:24] c=IN IP4 99.99.255.55 [May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0 [May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 48086 typ host generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 48086 typ srflx raddr 192.168.1.110 rport 48086 generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192
Re: [asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video
Hello is this mailing list still active ? Op 10-05-19 om 14:10 schreef Jonas Kellens: Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition : Video Support: Yes Prim.Transp. : WS Allowed.Trsp : WSS SIP Options : (none) Codecs : (opus|h264) Status : OK (75 ms) Useragent : SIP.js/0.12.0 Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss RTP Engine : asterisk Encryption : Yes RTCP Mux : Yes Video Support: Yes Prim.Transp. : WS Allowed.Trsp : WSS SIP Options : (none) Codecs : (opus|h264) Status : OK (47 ms) Useragent : SIP.js/0.12.0 Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss RTP Engine : asterisk Encryption : Yes RTCP Mux : Yes In general sip.conf I have : videosupport=yes disallow=all allow=alaw allow=opus allow=h264 When one peer makes a SIP INVITE for a video call, it is clear to me that the necessary codec information is present (this all looks fine to me) : (calling webRTC client) SIP Debugging Enabled for IP: 99.99.255.55 [May 10 10:45:24] [May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 ---> [May 10 10:45:24] INVITE sip:1...@wss.mydomain.tld SIP/2.0 [May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692 [May 10 10:45:24] Max-Forwards: 70 [May 10 10:45:24] To: [May 10 10:45:24] From: "WC User Chrome" ;tag=sdmbqkquhe [May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu [May 10 10:45:24] CSeq: 4132 INVITE [May 10 10:45:24] Contact: [May 10 10:45:24] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [May 10 10:45:24] Supported: outbound [May 10 10:45:24] User-Agent: SIP.js/0.12.0 [May 10 10:45:24] Content-Type: application/sdp [May 10 10:45:24] Content-Length: 5098 [May 10 10:45:24] [May 10 10:45:24] v=0 [May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1 [May 10 10:45:24] s=- [May 10 10:45:24] t=0 0 [May 10 10:45:24] a=group:BUNDLE audio video [May 10 10:45:24] a=msid-semantic: WMS I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E [May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 [May 10 10:45:24] c=IN IP4 99.99.255.55 [May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0 [May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 34197 typ host generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 34197 typ srflx raddr 192.168.1.110 rport 34197 generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192.168.1.110 9 typ host tcptype active generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=ice-ufrag:y8md [May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le [May 10 10:45:24] a=ice-options:trickle [May 10 10:45:24] a=fingerprint:sha-256 C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B [May 10 10:45:24] a=setup:actpass [May 10 10:45:24] a=mid:audio [May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level [May 10 10:45:24] a=sendrecv [May 10 10:45:24] a=rtcp-mux [May 10 10:45:24] a=rtpmap:111 opus/48000/2 [May 10 10:45:24] a=rtcp-fb:111 transport-cc [May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1 [May 10 10:45:24] a=rtpmap:103 ISAC/16000 [May 10 10:45:24] a=rtpmap:104 ISAC/32000 [May 10 10:45:24] a=rtpmap:9 G722/8000 [May 10 10:45:24] a=rtpmap:0 PCMU/8000 [May 10 10:45:24] a=rtpmap:8 PCMA/8000 [May 10 10:45:24] a=rtpmap:106 CN/32000 [May 10 10:45:24] a=rtpmap:105 CN/16000 [May 10 10:45:24] a=rtpmap:13 CN/8000 [May 10 10:45:24] a=rtpmap:110 telephone-event/48000 [May 10 10:45:24] a=rtpmap:112 telephone-event/32000 [May 10 10:45:24] a=rtpmap:113 telephone-event/16000 [May 10 10:45:24] a=rtpmap:126 telephone-event/8000 [May 10 10:45:24] a=ssrc:401971016 cname:cd1IocMPYzY4lNYJ [May 10 10:45:24] a=ssrc:401971016 msid:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E f8eee8bd-dd47-4c14-866d-07069cab255f [May 10 10:45:24] a=ssrc:401971016 mslabel:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E [May 10 10:45:24] a=ssrc:401971016 label:f8eee8bd-dd47-4c14-866d-07069cab255f [May 10 10:45:24] m=video 48086 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 127 122 125 107 108 109 124 [May 10 10:45:24] c=IN IP4 99.99.255.55 [May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0 [May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 48086 typ host generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 48086 typ srflx raddr 192.168.1.110 rport 48086 generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192
[asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video
Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition : Video Support: Yes Prim.Transp. : WS Allowed.Trsp : WSS SIP Options : (none) Codecs : (opus|h264) Status : OK (75 ms) Useragent : SIP.js/0.12.0 Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss RTP Engine : asterisk Encryption : Yes RTCP Mux : Yes Video Support: Yes Prim.Transp. : WS Allowed.Trsp : WSS SIP Options : (none) Codecs : (opus|h264) Status : OK (47 ms) Useragent : SIP.js/0.12.0 Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss RTP Engine : asterisk Encryption : Yes RTCP Mux : Yes In general sip.conf I have : videosupport=yes disallow=all allow=alaw allow=opus allow=h264 When one peer makes a SIP INVITE for a video call, it is clear to me that the necessary codec information is present (this all looks fine to me) : (calling webRTC client) SIP Debugging Enabled for IP: 99.99.255.55 [May 10 10:45:24] [May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 ---> [May 10 10:45:24] INVITE sip:1...@wss.mydomain.tld SIP/2.0 [May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692 [May 10 10:45:24] Max-Forwards: 70 [May 10 10:45:24] To: [May 10 10:45:24] From: "WC User Chrome" ;tag=sdmbqkquhe [May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu [May 10 10:45:24] CSeq: 4132 INVITE [May 10 10:45:24] Contact: [May 10 10:45:24] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [May 10 10:45:24] Supported: outbound [May 10 10:45:24] User-Agent: SIP.js/0.12.0 [May 10 10:45:24] Content-Type: application/sdp [May 10 10:45:24] Content-Length: 5098 [May 10 10:45:24] [May 10 10:45:24] v=0 [May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1 [May 10 10:45:24] s=- [May 10 10:45:24] t=0 0 [May 10 10:45:24] a=group:BUNDLE audio video [May 10 10:45:24] a=msid-semantic: WMS I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E [May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 [May 10 10:45:24] c=IN IP4 99.99.255.55 [May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0 [May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 34197 typ host generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 34197 typ srflx raddr 192.168.1.110 rport 34197 generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192.168.1.110 9 typ host tcptype active generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=ice-ufrag:y8md [May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le [May 10 10:45:24] a=ice-options:trickle [May 10 10:45:24] a=fingerprint:sha-256 C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B [May 10 10:45:24] a=setup:actpass [May 10 10:45:24] a=mid:audio [May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level [May 10 10:45:24] a=sendrecv [May 10 10:45:24] a=rtcp-mux [May 10 10:45:24] a=rtpmap:111 opus/48000/2 [May 10 10:45:24] a=rtcp-fb:111 transport-cc [May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1 [May 10 10:45:24] a=rtpmap:103 ISAC/16000 [May 10 10:45:24] a=rtpmap:104 ISAC/32000 [May 10 10:45:24] a=rtpmap:9 G722/8000 [May 10 10:45:24] a=rtpmap:0 PCMU/8000 [May 10 10:45:24] a=rtpmap:8 PCMA/8000 [May 10 10:45:24] a=rtpmap:106 CN/32000 [May 10 10:45:24] a=rtpmap:105 CN/16000 [May 10 10:45:24] a=rtpmap:13 CN/8000 [May 10 10:45:24] a=rtpmap:110 telephone-event/48000 [May 10 10:45:24] a=rtpmap:112 telephone-event/32000 [May 10 10:45:24] a=rtpmap:113 telephone-event/16000 [May 10 10:45:24] a=rtpmap:126 telephone-event/8000 [May 10 10:45:24] a=ssrc:401971016 cname:cd1IocMPYzY4lNYJ [May 10 10:45:24] a=ssrc:401971016 msid:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E f8eee8bd-dd47-4c14-866d-07069cab255f [May 10 10:45:24] a=ssrc:401971016 mslabel:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E [May 10 10:45:24] a=ssrc:401971016 label:f8eee8bd-dd47-4c14-866d-07069cab255f [May 10 10:45:24] m=video 48086 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 127 122 125 107 108 109 124 [May 10 10:45:24] c=IN IP4 99.99.255.55 [May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0 [May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 48086 typ host generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 48086 typ srflx raddr 192.168.1.110 rport 48086 generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192.168.1.110 9 typ host tcptype active generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=ice-ufrag:y8
Re: [asterisk-users] Spontaneous reboot due to MySQL lookups ?
Hello thank you for your answer. This does not happen all the time. It happens about once every 4 months. I just can not pinpoint WHEN exactly it occurs. I just see in the verbose logfile that it occurs after a MYSQL insert/update/delete statement. If Asterisk 13 handels MYSQL connections in a better way, then indeed I should look for upgrade. Kind regards. Op 05-10-18 om 01:25 schreef John Novack: As others have said, clearly it ISN'T "just working" or you would not have posted the question To state again, I am using Version 13, though a few minor revisions behind, with MySql, on CentOS 6 and have no rebooting or other MySql related issues Clearly you need to state in more detail what issues remain, once you migrate to AT LEAST 13.xx, and state your OS after becoming current with Asterisk, MySql and the OS I use MySql on every incoming call, and also maintain call detail records in MySql for every call, and it just simply works, and has for some time. Although I may be using it quite differently that you, it simply works. Is this a newly developing issue, or has it persisted for some time What if any changes have been made to the dialplan etc? Have you considered a strictly hardware issue? Memory? HD? MB?? The crystal ball is very cloudy on this one! John Novack Jonas Kellens wrote: Hello thank you for your answer. If I read your (and others) reaction correctly I can conclude that this is an Asterisk problem and not a problem of MySQL or dialplan logic ? You should know that the MySQL database is heavily questioned : mysql> show status like '%onn%'; +--++ | Variable_name | Value | +--++ | Aborted_connects | 469 | | Connections | 132762 | | Max_used_connections | 8 | | Ssl_client_connects | 0 | | Ssl_connect_renegotiates | 0 | | Ssl_finished_connects | 0 | | Threads_connected | 3 | +--++ 7 rows in set (0.00 sec) I stick to 1.8 because it just works. I had some issues with version 11 and 13 in the past. Regards Jonas. Op 04-10-18 om 17:49 schreef John Novack: Woefully out of date. You really need to put your efforts into at least a modest upgrade I use version 13 with MySql queries built into the dialplan on CentOs 6 and have NO such issues, either performance or any restart or reboot. It simply works I never used either 1.6 or 1.8, going from 1.4 to version 11, which did require some syntax changes to the dialplan. Given that even version 11 is EOL, you really need to put your efforts into doing the migration rather than tracking this one down JMO John Novack Jonas Kellens wrote: Hello using Asterisk 1.8.32. I notice that there is a spontaneous reboot of the Asterisk system from time to time. When I look in the logs (verbose file) I noticed that every time this occurs it's at a moment that there is a MySQL action, be it a lookup or an insert/update/delete. I must say I do have some MySQL queries that occur in my dialplan when a call comes in, to look up different actions to perform on this call. An idea how to overcome this problem ? Seems a "performance" issue, no ?! Is it better to have these MySQL queries to be done by an external script (like a php script that I call with the System()-command or a SHELL()-command) ? Here are some examples from the verbose file. [Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing [s@sub-GetAlertInfo:3] MYSQL("SIP/SipAgenT01-317d", "Connect connid localhost myuser mypwd myDB") in new stack [Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing [s@sub-GetAlertInfo:5] MYSQL("SIP/SipAgenT01-317d", "Query resultid 1 SELECT uri, callinfo FROM distringtone WHERE onoff='1'") in new stack [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/logger.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/asterisk.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] manager.c: [Aug 22 15:19:18] == Manager registered action DataGet [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/codecs.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] Asterisk Dynamic Loader Starting: [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/modules.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]
Re: [asterisk-users] Spontaneous reboot due to MySQL lookups ?
Hello thank you for your answer. If I read your (and others) reaction correctly I can conclude that this is an Asterisk problem and not a problem of MySQL or dialplan logic ? You should know that the MySQL database is heavily questioned : mysql> show status like '%onn%'; +--++ | Variable_name | Value | +--++ | Aborted_connects | 469 | | Connections | 132762 | | Max_used_connections | 8 | | Ssl_client_connects | 0 | | Ssl_connect_renegotiates | 0 | | Ssl_finished_connects | 0 | | Threads_connected | 3 | +--++ 7 rows in set (0.00 sec) I stick to 1.8 because it just works. I had some issues with version 11 and 13 in the past. Regards Jonas. Op 04-10-18 om 17:49 schreef John Novack: Woefully out of date. You really need to put your efforts into at least a modest upgrade I use version 13 with MySql queries built into the dialplan on CentOs 6 and have NO such issues, either performance or any restart or reboot. It simply works I never used either 1.6 or 1.8, going from 1.4 to version 11, which did require some syntax changes to the dialplan. Given that even version 11 is EOL, you really need to put your efforts into doing the migration rather than tracking this one down JMO John Novack Jonas Kellens wrote: Hello using Asterisk 1.8.32. I notice that there is a spontaneous reboot of the Asterisk system from time to time. When I look in the logs (verbose file) I noticed that every time this occurs it's at a moment that there is a MySQL action, be it a lookup or an insert/update/delete. I must say I do have some MySQL queries that occur in my dialplan when a call comes in, to look up different actions to perform on this call. An idea how to overcome this problem ? Seems a "performance" issue, no ?! Is it better to have these MySQL queries to be done by an external script (like a php script that I call with the System()-command or a SHELL()-command) ? Here are some examples from the verbose file. [Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing [s@sub-GetAlertInfo:3] MYSQL("SIP/SipAgenT01-317d", "Connect connid localhost myuser mypwd myDB") in new stack [Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing [s@sub-GetAlertInfo:5] MYSQL("SIP/SipAgenT01-317d", "Query resultid 1 SELECT uri, callinfo FROM distringtone WHERE onoff='1'") in new stack [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/logger.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/asterisk.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] manager.c: [Aug 22 15:19:18] == Manager registered action DataGet [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/codecs.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] Asterisk Dynamic Loader Starting: [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/modules.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/res_config_mysql.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] res_config_mysql.c: [Aug 22 15:19:18] == MySQL RealTime driver loaded. [Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] res_config_mysql.so => (MySQL RealTime Configuration Driver) [Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- Executing [s@sub-GetSipAccountdetails:3] MYSQL("SIP/SipAgenT01-4184", "Connect connid localhost myuser mypwd myDB") in new stack [Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- Executing [s@sub-GetSipAccountdetails:4] MYSQL("SIP/SipAgenT01-4184", "Query resultid 1 SELECT SIPusername, currstatus, available FROM tbl_SIP WHERE ID="800"") in new stack [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing '/etc/asterisk/logger.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Found [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing '/etc/asterisk/asterisk.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Found [Aug 22 16:23:32] VERBOSE[24309] manager.c: [Aug 22 16:23:32] == Manager registered action DataGet [Aug 22 16:23:32]
[asterisk-users] Spontaneous reboot due to MySQL lookups ?
Hello using Asterisk 1.8.32. I notice that there is a spontaneous reboot of the Asterisk system from time to time. When I look in the logs (verbose file) I noticed that every time this occurs it's at a moment that there is a MySQL action, be it a lookup or an insert/update/delete. I must say I do have some MySQL queries that occur in my dialplan when a call comes in, to look up different actions to perform on this call. An idea how to overcome this problem ? Seems a "performance" issue, no ?! Is it better to have these MySQL queries to be done by an external script (like a php script that I call with the System()-command or a SHELL()-command) ? Here are some examples from the verbose file. [Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing [s@sub-GetAlertInfo:3] MYSQL("SIP/SipAgenT01-317d", "Connect connid localhost myuser mypwd myDB") in new stack [Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing [s@sub-GetAlertInfo:5] MYSQL("SIP/SipAgenT01-317d", "Query resultid 1 SELECT uri, callinfo FROM distringtone WHERE onoff='1'") in new stack [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/logger.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/asterisk.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] manager.c: [Aug 22 15:19:18] == Manager registered action DataGet [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/codecs.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] Asterisk Dynamic Loader Starting: [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/modules.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing '/etc/asterisk/res_config_mysql.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found [Aug 22 15:19:18] VERBOSE[3306] res_config_mysql.c: [Aug 22 15:19:18] == MySQL RealTime driver loaded. [Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] res_config_mysql.so => (MySQL RealTime Configuration Driver) [Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- Executing [s@sub-GetSipAccountdetails:3] MYSQL("SIP/SipAgenT01-4184", "Connect connid localhost myuser mypwd myDB") in new stack [Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- Executing [s@sub-GetSipAccountdetails:4] MYSQL("SIP/SipAgenT01-4184", "Query resultid 1 SELECT SIPusername, currstatus, available FROM tbl_SIP WHERE ID="800"") in new stack [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing '/etc/asterisk/logger.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Found [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing '/etc/asterisk/asterisk.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Found [Aug 22 16:23:32] VERBOSE[24309] manager.c: [Aug 22 16:23:32] == Manager registered action DataGet [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing '/etc/asterisk/codecs.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Found [Aug 22 16:23:32] VERBOSE[24309] loader.c: [Aug 22 16:23:32] Asterisk Dynamic Loader Starting: [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing '/etc/asterisk/modules.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Found [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing '/etc/asterisk/res_config_mysql.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Found [Aug 22 16:23:32] VERBOSE[24309] res_config_mysql.c: [Aug 22 16:23:32] == MySQL RealTime driver loaded. [Aug 22 16:23:32] VERBOSE[24309] loader.c: [Aug 22 16:23:32] res_config_mysql.so => (MySQL RealTime Configuration Driver) [Oct 4 10:11:25] VERBOSE[4944] pbx.c: [Oct 4 10:11:25] -- Executing [s@sub-settings:16] MYSQL("SIP/SipAgenT01-08cb", "Connect connid localhost myuser mypwd myDB") in new stack [Oct 4 10:11:25] VERBOSE[4944] pbx.c: [Oct 4 10:11:25] -- Executing [s@sub-settings:17] MYSQL("SIP/SipAgenT01-08cb", "Query resultid 1 SELECT blockID from DID where DID=987654321") in new stack [Oct 4 10:11:29] VERBOSE[4961] config.c: [Oct 4 10:11:29] == Parsing '/etc/asterisk/asterisk.conf': [Oct 4 10:11:29] VERBOSE[4961] config.c: [Oct 4 10:11:29] == Found [Oct 4 10:11:29] VERBOSE[4961] manager.c: [Oct 4 10:11:29] == Manager registered action DataGet [Oct 4 10:11:29] VERBOSE[4961] config.c: [Oct 4 10:11:29] == Parsing '/etc/asterisk/codecs.conf': [Oct 4 10:11:
[asterisk-users] doing dnsmgr_lookup for
Hello list is there a way to limit the number of dns lookup's for 1 and the same host ? I see on Asterisk CLI a flood of : [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' [May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' I have several sip peer definitions (sip trunks) pointing at this same host. Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)
For those never getting a decend answer by the community on this mailinglist, I share my solution to my video problem : preferred_codec_only=no (I had this on 'yes') Op 26-06-17 om 14:43 schreef Jonas Kellens: Hello this is the debug output of a test video call. You see codec negotiation but at the end only alaw is chosen and gone is the video ! [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Our capabilities are 0x20010a (gsm|alaw|g729|h264) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: This call needs video offers! [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: ** Our capability: 0x2a (gsm|alaw|h264) Video flag: False Text flag: False [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: -- Done with adding codecs to SDP [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Done building SDP. Settling with this capability: 0x2a (gsm|alaw|h264) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Initializing initreq for method INVITE - callid 6ca7f3fb70b56f6f6f9373b776cd495d@11.22.33.44:5060 [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 0 [ 54]: INVITE sip:sipaccount12@192.168.1.111:50104;ob SIP/2.0 [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK54e24150;rport [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 3 [ 62]: From: "My Account" ;tag=as130bc3f0 [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 4 [ 45]: To: [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 5 [ 37]: Contact: [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 6 [ 61]: Call-ID: 6ca7f3fb70b56f6f6f9373b776cd495d@11.22.33.44:5060 [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 8 [ 21]: User-Agent: mydomain [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 9 [ 35]: Date: Mon, 26 Jun 2017 12:20:55 GMT [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 11 [ 19]: Supported: replaces [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 12 [ 42]: Alert-Info: <http://127.0.0.1>;info=intern [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp ... [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 11.22.33.44:5060;rport=5060;received=11.22.33.44;branch=z9hG4bK54e24150 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 2 [ 61]: Call-ID: 6ca7f3fb70b56f6f6f9373b776cd495d@11.22.33.44:5060 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 3 [ 62]: From: "My Account" ;tag=as130bc3f0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 4 [ 76]: To: ;tag=8a1f91570e9f434c9da9aca27ded7fb9 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 6 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 7 [ 50]: Contact: [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 10 [ 19]: Content-Length: 469 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 11 [ 0]: [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 0 [ 3]: v=0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 1 [ 46]: o=- 3707475663 3707475664 IN IP4 192.168.1.111 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 2 [ 9]: s=pjmedia [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 3 [ 8]: b=AS:352 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 4 [ 5]: t=0 0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 5 [ 9]: a=X-nat:0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 6 [ 26]: m=audio 4000 RTP/AVP 8 101 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 7 [ 22]: c=IN IP4 192.168.1.111 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 8 [ 12]: b=TIAS:64000 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 9 [ 32]: a=rtcp:4001 IN IP4 192.168.1.111 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 10 [ 10]: a=s
Re: [asterisk-users] BLF and Call Queues
Hello concerning this question of aug 2012, I am now using 1.8.32.2 and it seems that the code of app_queue.c has changed. The function ast_devstate_changed() is no longer used. Can anyone tell me what it is replaced with ? Kind regards Op 18-08-12 om 12:42 schreef Alec Davis: -Original Message- From: Alec Davis [mailto:siva...@paradise.net.nz] Sent: Saturday, 18 August 2012 10:36 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] BLF and Call Queues I've seen this post. That's why I thought it was possible. I'm using 1.8.11 What is the difference between this post and asterisk 1.8.11 ? The patch hasn't been accepted by the community, thus isn't in asterisk trunk or any asterisk branches. Alec Jonas; In case you had't seen it, the patch is available from review board https://reviewboard.asterisk.org/r/1619/ using the 'Download Diff' link at the top right of the review. Or directly form here https://reviewboard.asterisk.org/r/1619/diff/raw/ Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)
15]: a=fmtp:101 0-16 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 14 [ 23]: m=video 4002 RTP/AVP 99 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 15 [ 22]: c=IN IP4 192.168.1.111 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 16 [ 13]: b=TIAS:256000 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 17 [ 32]: a=rtcp:4003 IN IP4 192.168.1.111 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 18 [ 10]: a=sendrecv [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 19 [ 22]: a=rtpmap:99 H264/9 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 20 [ 55]: a=fmtp:99 profile-level-id=42000a; packetization-mode=0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: SIP response 200 to standard invite [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP o=- 3707475663 3707475664 IN IP4 192.168.1.111... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP s=pjmedia... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP b=AS:352... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] rtp_engine.c: Setting payload 8 based on m type on 0x7efde80a5930 [Jun 26 14:20:57] DEBUG[1932] rtp_engine.c: Setting payload 101 based on m type on 0x7efde80a5930 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 192.168.1.111... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP b=TIAS:64000... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4001 IN IP4 192.168.1.111... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] rtp_engine.c: Setting payload 99 based on m type on 0x7efde80a47f0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP c=IN IP4 192.168.1.111... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP b=TIAS:256000... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP a=rtcp:4003 IN IP4 192.168.1.111... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP a=sendrecv... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP a=rtpmap:99 H264/9... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP a=fmtp:99 profile-level-id=42000a; packetization-mode=0... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: We're settling with these formats: 0x8 (alaw) Op 21-04-17 om 16:33 schreef Derek Bolichowski: *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Friday, April 21, 2017 10:18 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264) Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. Why not try removing all codecs from the SIP Peer (deny all, allow only H264), unregister the peer, and try a video call again? If it works, try adding G711 back but keep H264 at the top of the priority. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Let's encrypt privkey : Specified certificate file could not be used
Hello James I am running asterisk as root, just to 'disable' all issues related to file rights. So this should not be the problem. Kind regards. Op 03-06-17 om 08:09 schreef James Cloos: "JK" == Jonas Kellens writes: JK> [Jun 2 14:29:28] ERROR[27360][C-0ae5]: res_rtp_asterisk.c:1441 JK> ast_rtp_dtls_set_configuration: Specified certificate file JK> '/etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' for RTP instance JK> '0x7f920c538a78' could not be used That error means that openssl's SSL_CTX_use_certificate_file() returned an error. The later error is just a result of that one. Does the uid/gid used for asterisk have access to the key? If the uid you use for asterisk is called asterisk, run this as root: su -c 'cat /etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' - asterisk If it fails, then the problem is permissions. You may need to alter the permissions on /etc/letsencrypt to allow non-root uids to access the symlinks and their targets. -JimC -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Let's encrypt privkey : Specified certificate file could not be used
Hello I get the following error when using our Let's Encrypt ssl certificate for webRTC calls : [Jun 2 14:29:28] == DTLS ECDH initialized (secp256r1), faster PFS enabled [Jun 2 14:29:28] ERROR[27360][C-0ae5]: res_rtp_asterisk.c:1441 ast_rtp_dtls_set_configuration: Specified certificate file '/etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' for RTP instance '0x7f920c538a78' could not be used [Jun 2 14:29:28] ERROR[27360][C-0ae5]: chan_sip.c:5941 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP instance '0x7f920c538a78' (ws.mydomain.tld is of course masked) Any idea why Asterisk has a problem with the certificate ? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to know a call is being transfered
Hello thank you for your answer. However this does not help me to know when a call is being transfered. My question is simple : if A calls B, and then B tranfers (unattened or attended) the call to C, how can I know this happens ?? I see it happening on the CLI, but how can I "catch" this, for example in the dialplan logic ? Or through AMI perhaps ? Kind regards. J. Op 29-05-17 om 10:16 schreef Jonathan H: Well, once you've upgraded to a version of Asterisk which didn't become "EOL - DO NOT USE - NO FIXES" (!) almost 2 years ago, then you might be able use logging which was introduced 5 years ago in Asterisk 11. Although the "transfers" section in the info below says it "can be a little tricky...". Read on! https://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging Call ID Logging (which has nothing to do with caller ID) is a new feature of Asterisk 11 intended to help administrators and support givers to more quickly understand problems that occur during the course of calls. Channels are now bound to call identifiers which can be shared among a number of channels, threads, and other consumers. Transfers Transfers can be a little tricky to follow with the call ID logging feature. As a general rule, an attended transfer will always result in a new call ID being made because a separate call must occur between the party that initiates the transfer and whatever extension is going to receive it. Once the attended transfer is completed, the channel that was transferred will use the Call ID created when the transferrer called the recipient. Blind transfers are slightly more variable. If a SIP peer 'peer1' calls another SIP peer 'peer2' via the dial application and peer2 blind transfers peer1 elsewhere, the call ID will persist. If on the other hand, peer1 blind transfers peer2 at this point a new call ID will be created. When peer1 transfers peer2, peer2 has a new channel created which enters the PBX for the first time, so it creates a new call ID. When peer1 is transferred, it simply resumes running PBX, so the call is still considered the same call. By setting the debug level to 3 for the channel internal API (channel_internal_api.c), all call ID settings for every channel will be logged and this may be able to help when trying to keep track of calls through multiple transfers. On 29 May 2017 at 08:17, Jonas Kellens wrote: Hello using Asterisk 1.8.32.3. What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ? So I can log this information. Kind regards. J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best way to know a call is being transfered
Hello using Asterisk 1.8.32.3. What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ? So I can log this information. Kind regards. J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)
Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. On 21-04-17 12:28, Marcelo Terres wrote: Did you try to activate DEBUG and set the verbosity to a higher level (100?) to check what Asterisk tells you about? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 20 April 2017 at 12:42, Jonas Kellens wrote: Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: I suppose that you enable the video support on sip.conf, right? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 19 April 2017 at 13:18, Jonas Kellens wrote: Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARY HEX TYPE NAME DESCRIPTION --- 1 (1 << 0)(0x1) audio g723 (G.723.1) 2 (1 << 1)(0x2) audiogsm (GSM) 4 (1 << 2)(0x4) audio ulaw (G.711 u-law) 8 (1 << 3)(0x8) audio alaw (G.711 A-law) 16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 << 5) (0x20) audio adpcm (ADPCM) 64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 << 7) (0x80) audio lpc10 (LPC10) 256 (1 << 8) (0x100) audio g729 (G.729A) 512 (1 << 9) (0x200) audio speex (SpeeX) 1024 (1 << 10) (0x400) audio ilbc (iLBC) 2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551) 4096 (1 << 12) (0x1000) audio g722 (G722) 8192 (1 << 13) (0x2000) audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom)) 16384 (1 << 14) (0x4000) audiosiren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom)) 32768 (1 << 15) (0x8000) audio slin16 (16 bit Signed Linear PCM (16kHz)) 65536 (1 << 16)(0x1) image jpeg (JPEG image) 131072 (1 << 17)(0x2) imagepng (PNG image) 262144 (1 << 18)(0x4) video h261 (H.261 Video) 524288 (1 << 19)(0x8) video h263 (H.263 Video) 1048576 (1 << 20) (0x10) video h263p (H.263+ Video) 2097152 (1 << 21) (0x20) video h264 (H.264 Video) 4194304 (1 << 22) (0x40) video mpeg4 (MPEG4 Video) 8388608 (1 << 23) (0x80) videounknown (unknown) 16777216 (1 << 24) (0x100) videounknown (unknown) 33554432 (1 << 25) (0x200) textunknown (unknown) 67108864 (1 << 26) (0x400) textred (T.140 Realtime Text with redundancy) 134217728 (1 << 27) (0x800) text t140 (Passthrough T.140 Realtime Text) 268435456 (1 << 28) (0x1000) textunknown (unknown) 536870912 (1 << 29) (0x2000) textunknown (unknown) 1073741824 (1 << 30) (0x4000) (unk)unknown (unknown) 2147483648 (1 << 31) (0x8000) (unk)unknown (unknown) 4294967296 (1 << 32)(0x1) audio g719 (ITU G.719) 8589934592 (1 << 33)(0x2) audiospeex16 (SpeeX 16khz) 17179869184 (1 << 34)(0x4) audiounknown (unknown) 34359738368 (1 << 35)(0x8) audiounknown (unknown) 68719476736 (1 << 36) (0x10) audiounknown (unknown) 137438953472 (1 << 37) (0x20) audiounknown (unknown) 274877906944 (1 << 38) (0x40) audioun
Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)
Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: I suppose that you enable the video support on sip.conf, right? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 19 April 2017 at 13:18, Jonas Kellens wrote: Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARY HEX TYPE NAME DESCRIPTION --- 1 (1 << 0)(0x1) audio g723 (G.723.1) 2 (1 << 1)(0x2) audiogsm (GSM) 4 (1 << 2)(0x4) audio ulaw (G.711 u-law) 8 (1 << 3)(0x8) audio alaw (G.711 A-law) 16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 << 5) (0x20) audio adpcm (ADPCM) 64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 << 7) (0x80) audio lpc10 (LPC10) 256 (1 << 8) (0x100) audio g729 (G.729A) 512 (1 << 9) (0x200) audio speex (SpeeX) 1024 (1 << 10) (0x400) audio ilbc (iLBC) 2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551) 4096 (1 << 12) (0x1000) audio g722 (G722) 8192 (1 << 13) (0x2000) audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom)) 16384 (1 << 14) (0x4000) audiosiren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom)) 32768 (1 << 15) (0x8000) audio slin16 (16 bit Signed Linear PCM (16kHz)) 65536 (1 << 16)(0x1) image jpeg (JPEG image) 131072 (1 << 17)(0x2) imagepng (PNG image) 262144 (1 << 18)(0x4) video h261 (H.261 Video) 524288 (1 << 19)(0x8) video h263 (H.263 Video) 1048576 (1 << 20) (0x10) video h263p (H.263+ Video) 2097152 (1 << 21) (0x20) video h264 (H.264 Video) 4194304 (1 << 22) (0x40) video mpeg4 (MPEG4 Video) 8388608 (1 << 23) (0x80) videounknown (unknown) 16777216 (1 << 24) (0x100) videounknown (unknown) 33554432 (1 << 25) (0x200) textunknown (unknown) 67108864 (1 << 26) (0x400) textred (T.140 Realtime Text with redundancy) 134217728 (1 << 27) (0x800) text t140 (Passthrough T.140 Realtime Text) 268435456 (1 << 28) (0x1000) textunknown (unknown) 536870912 (1 << 29) (0x2000) textunknown (unknown) 1073741824 (1 << 30) (0x4000) (unk)unknown (unknown) 2147483648 (1 << 31) (0x8000) (unk)unknown (unknown) 4294967296 (1 << 32)(0x1) audio g719 (ITU G.719) 8589934592 (1 << 33)(0x2) audiospeex16 (SpeeX 16khz) 17179869184 (1 << 34)(0x4) audiounknown (unknown) 34359738368 (1 << 35)(0x8) audiounknown (unknown) 68719476736 (1 << 36) (0x10) audiounknown (unknown) 137438953472 (1 << 37) (0x20) audiounknown (unknown) 274877906944 (1 << 38) (0x40) audiounknown (unknown) 549755813888 (1 << 39) (0x80) audiounknown (unknown) 1099511627776 (1 << 40) (0x100) audiounknown (unknown) 219902322 (1 << 41) (0x200) audiounknown (unknown) 4398046511104 (1 << 42) (0x400) audiounknown (unknown) 8796093022208 (1 << 43) (0x800) audiounknown (unknown) 17592186044416 (1 << 44) (0x1000) audiounknown (unknown)
[asterisk-users] Asterisk 1.8.32.3 : no video (h.264)
Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARY HEX TYPE NAME DESCRIPTION --- 1 (1 << 0)(0x1) audio g723 (G.723.1) 2 (1 << 1)(0x2) audiogsm (GSM) 4 (1 << 2)(0x4) audio ulaw (G.711 u-law) 8 (1 << 3)(0x8) audio alaw (G.711 A-law) 16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 << 5) (0x20) audio adpcm (ADPCM) 64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 << 7) (0x80) audio lpc10 (LPC10) 256 (1 << 8) (0x100) audio g729 (G.729A) 512 (1 << 9) (0x200) audio speex (SpeeX) 1024 (1 << 10) (0x400) audio ilbc (iLBC) 2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551) 4096 (1 << 12) (0x1000) audio g722 (G722) 8192 (1 << 13) (0x2000) audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom)) 16384 (1 << 14) (0x4000) audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom)) 32768 (1 << 15) (0x8000) audio slin16 (16 bit Signed Linear PCM (16kHz)) 65536 (1 << 16)(0x1) image jpeg (JPEG image) 131072 (1 << 17)(0x2) imagepng (PNG image) 262144 (1 << 18)(0x4) video h261 (H.261 Video) 524288 (1 << 19)(0x8) video h263 (H.263 Video) 1048576 (1 << 20) (0x10) video h263p (H.263+ Video) 2097152 (1 << 21) (0x20) video h264 (H.264 Video) 4194304 (1 << 22) (0x40) video mpeg4 (MPEG4 Video) 8388608 (1 << 23) (0x80) video unknown (unknown) 16777216 (1 << 24) (0x100) video unknown (unknown) 33554432 (1 << 25) (0x200) text unknown (unknown) 67108864 (1 << 26) (0x400) textred (T.140 Realtime Text with redundancy) 134217728 (1 << 27) (0x800) text t140 (Passthrough T.140 Realtime Text) 268435456 (1 << 28) (0x1000) text unknown (unknown) 536870912 (1 << 29) (0x2000) text unknown (unknown) 1073741824 (1 << 30) (0x4000) (unk) unknown (unknown) 2147483648 (1 << 31) (0x8000) (unk) unknown (unknown) 4294967296 (1 << 32)(0x1) audio g719 (ITU G.719) 8589934592 (1 << 33)(0x2) audio speex16 (SpeeX 16khz) 17179869184 (1 << 34)(0x4) audio unknown (unknown) 34359738368 (1 << 35)(0x8) audio unknown (unknown) 68719476736 (1 << 36) (0x10) audio unknown (unknown) 137438953472 (1 << 37) (0x20) audio unknown (unknown) 274877906944 (1 << 38) (0x40) audio unknown (unknown) 549755813888 (1 << 39) (0x80) audio unknown (unknown) 1099511627776 (1 << 40) (0x100) audio unknown (unknown) 219902322 (1 << 41) (0x200) audio unknown (unknown) 4398046511104 (1 << 42) (0x400) audio unknown (unknown) 8796093022208 (1 << 43) (0x800) audio unknown (unknown) 17592186044416 (1 << 44) (0x1000) audio unknown (unknown) 35184372088832 (1 << 45) (0x2000) audio unknown (unknown) 70368744177664 (1 << 46) (0x4000) audio unknown (unknown) 140737488355328 (1 << 47) (0x8000) audio testlaw (G.711 test-law) 281474976710656 (1 << 48)(0x1) video unknown (unknown) 562949953421312 (1 << 49)(0x2) video unknown (unknown) 1125899906842624 (1 << 50)(0x4) video unknown (unknown) 2251799813685248 (1 << 51)(0x8) video unknown (unknown) 4503599627370496 (1 << 52) (0x10) video unknown (unknown) 9007199254740992 (1 << 53) (0x20) video unknow
Re: [asterisk-users] Define SIP fromuser field in Dial()-command
Hello function sip_header is read-only. Kind regards. J. On 14-04-17 11:28, registrator wrote: In this case you will help function SIP_HEADER(from) Sent from: Lenovo P70-A On Apr 14, 2017 12:04 PM, Jonas Kellens wrote: Hello this does not set user field in From-header. I get : From: "user762" ;tag=as7f44c043 What I want is : From: "9876543210" ;tag=as7f44c043 I need this part : you see the user part ? I need to set the value 'user762' Kind regards J. On 14-04-17 10:46, registrator wrote: Hello! May be you help CALLERID(name) function? exten => _X.,1,Set(CALLERID(name)=$name) Then you well see INVITE SIP : FROM "$name" . Sent from: Lenovo P70-A On Apr 14, 2017 10:54 AM, Jonas Kellens wrote: Hello any input on this ? How to set user-field in From-header with the Dial()-command in dialplan ? Kind regards J. On 03-04-17 10:25, Jonas Kellens wrote: Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN}) The From part of the SIP INVITE always has the EXTEN in it in stead of the user (user762) : From: "the_extension" ;tag=as224453ac How can I get : From: "the_extension" ;tag=as224453ac ?? I know about sip.conf. That is not the question. My question is clear : how to set this in dialplan ? Thank you for the feedback. Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Define SIP fromuser field in Dial()-command
Hello this does not set user field in From-header. I get : From: "user762" ;tag=as7f44c043 What I want is : From: "9876543210" ;tag=as7f44c043 I need this part : you see the user part ? I need to set the value 'user762' Kind regards J. On 14-04-17 10:46, registrator wrote: Hello! May be you help CALLERID(name) function? exten => _X.,1,Set(CALLERID(name)=$name) Then you well see INVITE SIP : FROM "$name" . Sent from: Lenovo P70-A On Apr 14, 2017 10:54 AM, Jonas Kellens wrote: Hello any input on this ? How to set user-field in From-header with the Dial()-command in dialplan ? Kind regards J. On 03-04-17 10:25, Jonas Kellens wrote: Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN}) The From part of the SIP INVITE always has the EXTEN in it in stead of the user (user762) : From: "the_extension";tag=as224453ac How can I get : From: "the_extension";tag=as224453ac ?? I know about sip.conf. That is not the question. My question is clear : how to set this in dialplan ? Thank you for the feedback. Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Define SIP fromuser field in Dial()-command
Hello any input on this ? How to set user-field in From-header with the Dial()-command in dialplan ? Kind regards J. On 03-04-17 10:25, Jonas Kellens wrote: Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN}) The From part of the SIP INVITE always has the EXTEN in it in stead of the user (user762) : From: "the_extension" ;tag=as224453ac How can I get : From: "the_extension" ;tag=as224453ac ?? I know about sip.conf. That is not the question. My question is clear : how to set this in dialplan ? Thank you for the feedback. Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Define SIP fromuser field in Dial()-command
Hello in what way does this set the 'fromuser' field in the SIP INVITE ? Kind regards. J. On 05-04-17 22:05, Pete Mundy wrote: Hi Jonas Does the information at this link help? http://the-asterisk-book.com/1.6/funktionen-callerid.html Pete On 5/04/2017, at 8:11 pm, Jonas Kellens <mailto:jonas.kell...@telenet.be>> wrote: Hello anyone have some useful input on this ? Thanks. On 03-04-17 10:25, Jonas Kellens wrote: Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Define SIP fromuser field in Dial()-command
Hello anyone have some useful input on this ? Thanks. On 03-04-17 10:25, Jonas Kellens wrote: Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN}) The From part of the SIP INVITE always has the EXTEN in it in stead of the user (user762) : From: "the_extension" ;tag=as224453ac How can I get : From: "the_extension" ;tag=as224453ac ?? I know about sip.conf. That is not the question. My question is clear : how to set this in dialplan ? Thank you for the feedback. Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Define SIP fromuser field in Dial()-command
Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN}) The From part of the SIP INVITE always has the EXTEN in it in stead of the user (user762) : From: "the_extension" ;tag=as224453ac How can I get : From: "the_extension" ;tag=as224453ac ?? I know about sip.conf. That is not the question. My question is clear : how to set this in dialplan ? Thank you for the feedback. Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moh reload not reloading/reading new musiconhold files
Hello I can confirm that touch-ing /etc/asterisk/musiconhold.conf (just open with vi and close again) and then issuing a 'module reload res_musiconhold.so' on the Asterisk CLI makes the new files load into Asterisk. Very strange !! I would not know how to automate this through script... Kind regards. On 24-03-17 12:29, Daniel Journo wrote: > Hello > as you can read in my original post "moh reload" and "module reload res_musiconhold.so" does nothing. > Only at restart the new files are available. > Is this a bug ?? How can I get more debugging for this problem ?? I think there is currently a bug with MOH. For now, if you add a file to a moh folder, ‘touch musiconhold.conf’ and then reload moh. Please let me know how it goes. Kind regards Dan Journo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moh reload not reloading/reading new musiconhold files
Hello as you can read in my original post "moh reload" and "module reload res_musiconhold.so" does nothing. Only at restart the new files are available. Is this a bug ?? How can I get more debugging for this problem ?? Kind regards. On 23-03-17 22:54, Administrator TOOTAI wrote: Le 23/03/2017 à 20:17, Jonas Kellens a écrit : Hello is there any more information on how to reload/read musiconhold files ? CLI> module reload res_musiconhold -- Daniel On 07-03-17 10:46, Jonas Kellens wrote: Hello I did not mention it but of course the MOH directory is listed in /etc/asterisk/musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh [myfolder_1] mode=files directory=/var/lib/asterisk/moh/myfolder/1 sort=alpha [myfolder_2] mode=files directory=/var/lib/asterisk/moh/myfolder/2 sort=alpha [myfolder_3] mode=files directory=/var/lib/asterisk/moh/myfolder/3 sort=alpha No mather where I put the new file, it is never listed. Untill a full restart of Asterisk ! Then it is listed. But is there no other way to load/read a new MOH file than to completely restart Asterisk ?? After Asterisk restart : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/macroform-the_simplicity File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day Class: myfolder_1 File: /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity Kind regards. On 03-03-17 18:26, John Kiniston wrote: Your new file is in the 'myfolder/1'' subdirectory of the MOH directory. Either move the file into the MOH directory or define a new class in musiconhold.conf that is for your directory. On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello using Asterisk 1.8.32.3 Current music on hold : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity New musiconhold file : [root@myserver ]# file /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz I issue a reload of the moh : myserver*CLI> moh reload myserver*CLI> module reload res_musiconhold.so [Mar 3 15:04:53] -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) Situation after reload : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity Even a complete 'module reload' on Asterisk CLI does nothing : myserver*CLI> module reload [Mar 3 15:13:54] == Parsing '/etc/asterisk/extconfig.conf': [Mar 3 15:13:54] == Found [Mar 3 15:13:54] == Parsing '/etc/asterisk/logger.conf': [Mar 3 15:13:54] == Found ... [Mar 3 15:13:54] -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) ... Situation after reload : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity So, reloading musiconhold does not reload/read musiconhold files. How to read/load new musiconhold files into asterisk ?? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ <https://community.asterisk.org/> New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started <https://wiki.asterisk.org/wiki/display/AST/Getting+Started> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts
Re: [asterisk-users] moh reload not reloading/reading new musiconhold files
Hello is there any more information on how to reload/read musiconhold files ? Kind regards. On 07-03-17 10:46, Jonas Kellens wrote: Hello I did not mention it but of course the MOH directory is listed in /etc/asterisk/musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh [myfolder_1] mode=files directory=/var/lib/asterisk/moh/myfolder/1 sort=alpha [myfolder_2] mode=files directory=/var/lib/asterisk/moh/myfolder/2 sort=alpha [myfolder_3] mode=files directory=/var/lib/asterisk/moh/myfolder/3 sort=alpha No mather where I put the new file, it is never listed. Untill a full restart of Asterisk ! Then it is listed. But is there no other way to load/read a new MOH file than to completely restart Asterisk ?? After Asterisk restart : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/macroform-the_simplicity File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day Class: myfolder_1 File: /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity Kind regards. On 03-03-17 18:26, John Kiniston wrote: Your new file is in the 'myfolder/1'' subdirectory of the MOH directory. Either move the file into the MOH directory or define a new class in musiconhold.conf that is for your directory. On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello using Asterisk 1.8.32.3 Current music on hold : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity New musiconhold file : [root@myserver ]# file /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz I issue a reload of the moh : myserver*CLI> moh reload myserver*CLI> module reload res_musiconhold.so [Mar 3 15:04:53] -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) Situation after reload : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity Even a complete 'module reload' on Asterisk CLI does nothing : myserver*CLI> module reload [Mar 3 15:13:54] == Parsing '/etc/asterisk/extconfig.conf': [Mar 3 15:13:54] == Found [Mar 3 15:13:54] == Parsing '/etc/asterisk/logger.conf': [Mar 3 15:13:54] == Found ... [Mar 3 15:13:54] -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) ... Situation after reload : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity So, reloading musiconhold does not reload/read musiconhold files. How to read/load new musiconhold files into asterisk ?? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ <https://community.asterisk.org/> New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started <https://wiki.asterisk.org/wiki/display/AST/Getting+Started> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- ___
Re: [asterisk-users] moh reload not reloading/reading new musiconhold files
Hello I did not mention it but of course the MOH directory is listed in /etc/asterisk/musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh [myfolder_1] mode=files directory=/var/lib/asterisk/moh/myfolder/1 sort=alpha [myfolder_2] mode=files directory=/var/lib/asterisk/moh/myfolder/2 sort=alpha [myfolder_3] mode=files directory=/var/lib/asterisk/moh/myfolder/3 sort=alpha No mather where I put the new file, it is never listed. Untill a full restart of Asterisk ! Then it is listed. But is there no other way to load/read a new MOH file than to completely restart Asterisk ?? After Asterisk restart : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/macroform-the_simplicity File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day Class: myfolder_1 File: /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity Kind regards. On 03-03-17 18:26, John Kiniston wrote: Your new file is in the 'myfolder/1'' subdirectory of the MOH directory. Either move the file into the MOH directory or define a new class in musiconhold.conf that is for your directory. On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello using Asterisk 1.8.32.3 Current music on hold : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity New musiconhold file : [root@myserver ]# file /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz I issue a reload of the moh : myserver*CLI> moh reload myserver*CLI> module reload res_musiconhold.so [Mar 3 15:04:53] -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) Situation after reload : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity Even a complete 'module reload' on Asterisk CLI does nothing : myserver*CLI> module reload [Mar 3 15:13:54] == Parsing '/etc/asterisk/extconfig.conf': [Mar 3 15:13:54] == Found [Mar 3 15:13:54] == Parsing '/etc/asterisk/logger.conf': [Mar 3 15:13:54] == Found ... [Mar 3 15:13:54] -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) ... Situation after reload : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity So, reloading musiconhold does not reload/read musiconhold files. How to read/load new musiconhold files into asterisk ?? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ <https://community.asterisk.org/> New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started <https://wiki.asterisk.org/wiki/display/AST/Getting+Started> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk
[asterisk-users] moh reload not reloading/reading new musiconhold files
Hello using Asterisk 1.8.32.3 Current music on hold : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity New musiconhold file : [root@myserver ]# file /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz I issue a reload of the moh : myserver*CLI> moh reload myserver*CLI> module reload res_musiconhold.so [Mar 3 15:04:53] -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) Situation after reload : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity Even a complete 'module reload' on Asterisk CLI does nothing : myserver*CLI> module reload [Mar 3 15:13:54] == Parsing '/etc/asterisk/extconfig.conf': [Mar 3 15:13:54] == Found [Mar 3 15:13:54] == Parsing '/etc/asterisk/logger.conf': [Mar 3 15:13:54] == Found ... [Mar 3 15:13:54] -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) ... Situation after reload : myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity So, reloading musiconhold does not reload/read musiconhold files. How to read/load new musiconhold files into asterisk ?? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour
On 21-11-16 17:20, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: On 21-11-16 15:17, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-0246' [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack [Nov 21 08:18:26] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-003c;2", "SIP/mysip692") in new stack [Nov 21 08:18:26] app_dial.c: Called SIP/mysip692 Example 2 : [Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack [Nov 21 08:20:11] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-0255' [Nov 21 08:20:45] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-0040;2", "SIP/mysip692") in new stack [Nov 21 08:20:45] app_dial.c: Called SIP/mysip692 I did not see this behaviour in previous Asterisk versions. Could this be a bug ? There's not enough information here to know what is preventing the call from occurring. I'd look at a debug log between the caller entering the Queue and the outbound call being made. That should illustrate what is causing the delay. -- Matthew Jordan Hello and what exactly am I looking for in the debug logs ? I have generated debug output and re-produced the issue. Again 23 seconds before calling the queue member : [Nov 21 16:23:33] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-4e6e", "myqueue1300,,,") in new stack [Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-4e6e' [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-081a;2", "") in new stack [Nov 21 16:23:56] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:2] NoOp("Local/mysip692@CallFromQueue-081a;2", "exten = mysip692") in new stack [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-081a;2", "SIP/mysip692") in new stack [Nov 21 16:23:56] app_dial.c: Called SIP/mysip692 [Nov 21 16:23:56] app_dial.c: SIP/mysip692-4e86 is ringing [Nov 21 16:23:56] app_queue.c: Local/mysip692@CallFromQueue-081a;1 is ringing Could it be that it is because my Queue member 'mysip692' is occupied in another bridge (call) ? This I see in the logs just before the Call Queue starts calling the queue member : [Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged 'SIP/mysip-4e6a' and 'SIP/incoming-4e63' in stack [Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-4e63 left 'native_rtp' basic-bridge [Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-4e6a left 'native_rtp' basic-bridge A bit too coincidal, no ? So then it has something to do with the bridging ? I did not have this behaviour in previous Asterisk versions. Those aren't debug logs. Instructions for generating debug information can be found on the wiki: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information That being said, if the Queue Member is currently busy (which will be denoted by their device state), and you have not configured the Queue to ring the Queue Member while they are busy, then I would expect any new caller to hang out in the Queue until that Member is available. -- Matthew
Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour
On 21-11-16 19:14, Jonas Kellens wrote: On 21-11-16 17:20, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: On 21-11-16 15:17, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-0246' [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack [Nov 21 08:18:26] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-003c;2", "SIP/mysip692") in new stack [Nov 21 08:18:26] app_dial.c: Called SIP/mysip692 Example 2 : [Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack [Nov 21 08:20:11] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-0255' [Nov 21 08:20:45] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-0040;2", "SIP/mysip692") in new stack [Nov 21 08:20:45] app_dial.c: Called SIP/mysip692 I did not see this behaviour in previous Asterisk versions. Could this be a bug ? There's not enough information here to know what is preventing the call from occurring. I'd look at a debug log between the caller entering the Queue and the outbound call being made. That should illustrate what is causing the delay. -- Matthew Jordan Hello and what exactly am I looking for in the debug logs ? I have generated debug output and re-produced the issue. Again 23 seconds before calling the queue member : [Nov 21 16:23:33] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-4e6e", "myqueue1300,,,") in new stack [Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-4e6e' [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-081a;2", "") in new stack [Nov 21 16:23:56] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:2] NoOp("Local/mysip692@CallFromQueue-081a;2", "exten = mysip692") in new stack [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-081a;2", "SIP/mysip692") in new stack [Nov 21 16:23:56] app_dial.c: Called SIP/mysip692 [Nov 21 16:23:56] app_dial.c: SIP/mysip692-4e86 is ringing [Nov 21 16:23:56] app_queue.c: Local/mysip692@CallFromQueue-081a;1 is ringing Could it be that it is because my Queue member 'mysip692' is occupied in another bridge (call) ? This I see in the logs just before the Call Queue starts calling the queue member : [Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged 'SIP/mysip-4e6a' and 'SIP/incoming-4e63' in stack [Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-4e63 left 'native_rtp' basic-bridge [Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-4e6a left 'native_rtp' basic-bridge A bit too coincidal, no ? So then it has something to do with the bridging ? I did not have this behaviour in previous Asterisk versions. Those aren't debug logs. Instructions for generating debug information can be found on the wiki: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information That being said, if the Queue Member is currently busy (which will be denoted by their device state), and you have not configured the Queue to ring the Queue Member while they are busy, then I would expect any
Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour
On 21-11-16 17:20, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: On 21-11-16 15:17, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-0246' [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack [Nov 21 08:18:26] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-003c;2", "SIP/mysip692") in new stack [Nov 21 08:18:26] app_dial.c: Called SIP/mysip692 Example 2 : [Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack [Nov 21 08:20:11] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-0255' [Nov 21 08:20:45] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-0040;2", "SIP/mysip692") in new stack [Nov 21 08:20:45] app_dial.c: Called SIP/mysip692 I did not see this behaviour in previous Asterisk versions. Could this be a bug ? There's not enough information here to know what is preventing the call from occurring. I'd look at a debug log between the caller entering the Queue and the outbound call being made. That should illustrate what is causing the delay. -- Matthew Jordan Hello and what exactly am I looking for in the debug logs ? I have generated debug output and re-produced the issue. Again 23 seconds before calling the queue member : [Nov 21 16:23:33] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-4e6e", "myqueue1300,,,") in new stack [Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-4e6e' [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-081a;2", "") in new stack [Nov 21 16:23:56] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:2] NoOp("Local/mysip692@CallFromQueue-081a;2", "exten = mysip692") in new stack [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-081a;2", "SIP/mysip692") in new stack [Nov 21 16:23:56] app_dial.c: Called SIP/mysip692 [Nov 21 16:23:56] app_dial.c: SIP/mysip692-4e86 is ringing [Nov 21 16:23:56] app_queue.c: Local/mysip692@CallFromQueue-081a;1 is ringing Could it be that it is because my Queue member 'mysip692' is occupied in another bridge (call) ? This I see in the logs just before the Call Queue starts calling the queue member : [Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged 'SIP/mysip-4e6a' and 'SIP/incoming-4e63' in stack [Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-4e63 left 'native_rtp' basic-bridge [Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-4e6a left 'native_rtp' basic-bridge A bit too coincidal, no ? So then it has something to do with the bridging ? I did not have this behaviour in previous Asterisk versions. Those aren't debug logs. Instructions for generating debug information can be found on the wiki: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information That being said, if the Queue Member is currently busy (which will be denoted by their device state), and you have not configured the Queue to ring the Queue Member while they are busy, then I would expect any new caller to hang out in the Queue until that Member is available. -- Matthew Jord
Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour
On 21-11-16 15:17, Matthew Jordan wrote: On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-0246' [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack [Nov 21 08:18:26] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-003c;2", "SIP/mysip692") in new stack [Nov 21 08:18:26] app_dial.c: Called SIP/mysip692 Example 2 : [Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack [Nov 21 08:20:11] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-0255' [Nov 21 08:20:45] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-0040;2", "SIP/mysip692") in new stack [Nov 21 08:20:45] app_dial.c: Called SIP/mysip692 I did not see this behaviour in previous Asterisk versions. Could this be a bug ? There's not enough information here to know what is preventing the call from occurring. I'd look at a debug log between the caller entering the Queue and the outbound call being made. That should illustrate what is causing the delay. -- Matthew Jordan Hello and what exactly am I looking for in the debug logs ? I have generated debug output and re-produced the issue. Again 23 seconds before calling the queue member : [Nov 21 16:23:33] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-4e6e", "myqueue1300,,,") in new stack [Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-4e6e' [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-081a;2", "") in new stack [Nov 21 16:23:56] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:2] NoOp("Local/mysip692@CallFromQueue-081a;2", "exten = mysip692") in new stack [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-081a;2", "SIP/mysip692") in new stack [Nov 21 16:23:56] app_dial.c: Called SIP/mysip692 [Nov 21 16:23:56] app_dial.c: SIP/mysip692-4e86 is ringing [Nov 21 16:23:56] app_queue.c: Local/mysip692@CallFromQueue-081a;1 is ringing Could it be that it is because my Queue member 'mysip692' is occupied in another bridge (call) ? This I see in the logs just before the Call Queue starts calling the queue member : [Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged 'SIP/mysip-4e6a' and 'SIP/incoming-4e63' in stack [Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-4e63 left 'native_rtp' basic-bridge [Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-4e6a left 'native_rtp' basic-bridge A bit too coincidal, no ? So then it has something to do with the bridging ? I did not have this behaviour in previous Asterisk versions. Kind regards. J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13.12.2 : strange queue behaviour
Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-0246' [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack [Nov 21 08:18:26] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-003c;2", "SIP/mysip692") in new stack [Nov 21 08:18:26] app_dial.c: Called SIP/mysip692 Example 2 : [Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15] Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack [Nov 21 08:20:11] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-0255' [Nov 21 08:20:45] app_queue.c: Called Local/mysip692@CallFromQueue [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQueue-0040;2", "SIP/mysip692") in new stack [Nov 21 08:20:45] app_dial.c: Called SIP/mysip692 I did not see this behaviour in previous Asterisk versions. Could this be a bug ? Kind regards. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload'
On 11-10-16 14:44, Joshua Colp wrote: Jonas Kellens wrote: Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip reload'. This doesn't ring a bell on any issues filed or any posts anywhere. Getting a backtrace[1] would show precisely where it is hanging though. Is it possible a host is having DNS issues? That can cause chan_sip to lock up for a period of time. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock Hello backtrace was generated and issue is reported : https://issues.asterisk.org/jira/browse/ASTERISK-26585 Kind regards. J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime queue & agent groups
Hello any one have some input on this ? I've already tried changing the membername to : testacc77000/@1 Is completely ignored. I've already tried changing the interface to : testacc77000/@1 Is completely ignored. Or is it just not possible to group queue members ?? Thanks. J. On 27-10-16 15:53, Jonas Kellens wrote: Hello I'm a bit confused on how to group agents (give agents a group number) when using realtime queues. I read on the wiki : * If you include groups in your queue definition the calls get routed in the order of the group regardless of the specified strategy. So I just have a member= line for each agent. member => Agent/@1 ; a group member => Agent/501 ; a single agent member => Agent/:1,1 ; Any agent in group 1, wait for first available, but consider with penalty In my realtime database I have table queue_members : +--++-++-+-++ | uniqueid | membername | queue_name | interface | state_interface | penalty | paused | +--++-++-+-++ | 2916 | testacc77000 | queue7700q4 | testacc77000 | | 0 | NULL | | 2917 | testacc77001 | queue7700q4 | testacc77001 | | 3 | NULL | | 2843 | testacc77000 | queue7700q4 | testacc77000 | | 0 | NULL | | 2905 | testacc7700905 | queue7700q5 | testacc7700905 | | 0 | NULL | | 2888 | testacc77000 | queue7700q5 | testacc77000 | | 0 | NULL | | 2900 | testacc77000 | queue7700q5 | testacc77000 | | 0 | NULL | | 2901 | testacc77001 | queue7700q5 | testacc77001 | | 0 | NULL | How do I define a group to a certain agent/member in this case ? Kind regards J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem setting up ssl connection
On 26-10-16 23:24, Stefan Tichy wrote: On Wed, Oct 26, 2016 at 04:57:15PM +0200, Jonas Kellens wrote: if it is indeed manager.conf that I need to edit then the problem is that I see no param : tlsdontverifyserver=yes A comment copied from sip.conf.sample: "If set to yes, don't verify the servers certificate when acting as a client." With AMI connections asterisk is allways the server. I don't know how to make the AMI ignore the self-signed certificate. The client fails to verfify the certificate. Do you use PHP 5.6? The default behavior has changed. Hello I use PHP 5.6.27. So I should be looking inside php.ini ? Kind regards J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime queue & agent groups
Hello I'm a bit confused on how to group agents (give agents a group number) when using realtime queues. I read on the wiki : * If you include groups in your queue definition the calls get routed in the order of the group regardless of the specified strategy. So I just have a member= line for each agent. member => Agent/@1 ; a group member => Agent/501 ; a single agent member => Agent/:1,1 ; Any agent in group 1, wait for first available, but consider with penalty In my realtime database I have table queue_members : +--++-++-+-++ | uniqueid | membername | queue_name | interface | state_interface | penalty | paused | +--++-++-+-++ | 2916 | testacc77000 | queue7700q4 | testacc77000 | | 0 | NULL | | 2917 | testacc77001 | queue7700q4 | testacc77001 | | 3 | NULL | | 2843 | testacc77000 | queue7700q4 | testacc77000 | | 0 | NULL | | 2905 | testacc7700905 | queue7700q5 | testacc7700905 | | 0 | NULL | | 2888 | testacc77000 | queue7700q5 | testacc77000 | | 0 | NULL | | 2900 | testacc77000 | queue7700q5 | testacc77000 | | 0 | NULL | | 2901 | testacc77001 | queue7700q5 | testacc77001 | | 0 | NULL | How do I define a group to a certain agent/member in this case ? Kind regards J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem setting up ssl connection
On 26-10-16 15:03, Dan Jenkins wrote: On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello I keep getting the following error when trying to connect to the Asterisk server using AMI : $socket = fsockopen("tls://11.22.33.44 <http://11.22.33.44>","5039", $errno, $errstr, 5); Erorr on CLI : [Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection: Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [Oct 26 14:38:19] WARNING[2992]: tcptls.c:684 handle_tcptls_connection: FILE * open failed! I have in sip.conf : tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlsdontverifyserver=yes tlscipher=ALL ;tlsclientmethod=tlsv2 /etc/asterisk/keys : -rw--- 1 root root 1,2K okt 26 14:25 asterisk.crt -rw--- 1 root root 574 okt 26 14:24 asterisk.csr -rw--- 1 root root 887 okt 26 14:24 asterisk.key -rw--- 1 root root 2,1K okt 26 14:25 asterisk.pem -rw--- 1 root root 160 okt 26 14:24 ca.cfg -rw--- 1 root root 1,8K okt 26 14:24 ca.crt -rw--- 1 root root 3,3K okt 26 14:24 ca.key -rw--- 1 root root 123 okt 26 14:24 tmp.cfg The webserver ( A ) from where I open the socket to tls://11.22.33.44 <http://11.22.33.44> also has a self-signed certificate. This problem started when creating a new self-signed cert on webserver A. Any thoughts ? Thanks ! Kind regards. J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ <https://community.asterisk.org/> New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started <https://wiki.asterisk.org/wiki/display/AST/Getting+Started> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> Jonas, You talk about sip.conf and setting your TLS cert there - but you're trying to connect to the AMI over TLS - so you need to set this stuff in manager.conf (https://github.com/asterisk/asterisk/blob/master/configs/samples/manager.conf.sample) - did you mean manager.conf ? The error says that it doesn't understand the Certificate Authority in the cert. The box you're connecting from shouldn't affect anything so the issue will be with the CA of the cert - usually you need to add the CA to the cert to complete the chain. If this is a public box then I'd recommend just using LetsEncrypt - many things don't like Self Signed Certs now Dan Hello Dan if it is indeed manager.conf that I need to edit then the problem is that I see no param : tlsdontverifyserver=yes I don't know how to make the AMI ignore the self-signed certificate. Kind regards J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem setting up ssl connection
Hello I keep getting the following error when trying to connect to the Asterisk server using AMI : $socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5); Erorr on CLI : [Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection: Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [Oct 26 14:38:19] WARNING[2992]: tcptls.c:684 handle_tcptls_connection: FILE * open failed! I have in sip.conf : tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlsdontverifyserver=yes tlscipher=ALL ;tlsclientmethod=tlsv2 /etc/asterisk/keys : -rw--- 1 root root 1,2K okt 26 14:25 asterisk.crt -rw--- 1 root root 574 okt 26 14:24 asterisk.csr -rw--- 1 root root 887 okt 26 14:24 asterisk.key -rw--- 1 root root 2,1K okt 26 14:25 asterisk.pem -rw--- 1 root root 160 okt 26 14:24 ca.cfg -rw--- 1 root root 1,8K okt 26 14:24 ca.crt -rw--- 1 root root 3,3K okt 26 14:24 ca.key -rw--- 1 root root 123 okt 26 14:24 tmp.cfg The webserver ( A ) from where I open the socket to tls://11.22.33.44 also has a self-signed certificate. This problem started when creating a new self-signed cert on webserver A. Any thoughts ? Thanks ! Kind regards. J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload'
Hello Edilson I have removed all sip peer definition in sip.conf. I only have rtcachefriends=yes for my mysql realtime sip peers (IP-phones). But what I don't understand is why I have no issue when for example using asterisk 1.8.32.3 again. Kind regards. On 11-10-16 14:50, Edilson Amaral wrote: Hi This happens to me when one peer (provider) is bad ! Try to remove all peers from your sip.conf and gradually add them back! *From:* Jonas Kellens *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Sent:* Tuesday, October 11, 2016 8:41 AM *Subject:* [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload' Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip reload'. This is what I have on CLI : sip5*CLI> sip reload [Oct 7 23:58:40] Reloading SIP [Oct 7 23:58:40] == Parsing '/etc/asterisk/sip.conf': Found [Oct 7 23:58:40] == Parsing '/etc/asterisk/sipTemplates.conf': Found [Oct 7 23:58:40] == Parsing '/etc/asterisk/users.conf': Found [Oct 7 23:58:40] == Using SIP TOS bits 96 [Oct 7 23:58:40] == Using SIP CoS mark 3 [Oct 7 23:58:40] == TLS/SSL ECDH initialized (secp256r1), faster PFS cipher-suites enabled [Oct 7 23:58:40] == TLS/SSL certificate ok --> no more output on CLI. Asterisk has gone completely ! Another 'sip reload' gives : sip5*CLI> sip reload [Oct 8 00:01:10] Previous SIP reload not yet done sip5*CLI> sip reload sip5*CLI> Other commands are no problem on the CLI (while the freeze occurs ! ) : sip5*CLI> core show version Asterisk certified/13.8-cert3 built by root @ sip5.mydomain.tld on a x86_64 running Linux on 2016-10-07 21:27:15 UTC sip5*CLI> sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 0 active SIP channels sip5*CLI> core show threads 0x7f97ff0fb700 2849 netconsole started at [ 1639] asterisk.c listener() 0x7f97fe843700 2760 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff367700 2759 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97fe8bf700 2758 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97fe93b700 2173 monitor_sig_flagsstarted at [ 4768] asterisk.c asterisk_daemon() 0x7f97fe9b7700 2172 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97fea33700 2171 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97feaaf700 2170 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97feb2b700 2169 scan_thread started at [ 920] pbx_spool.c load_module() 0x7f97feba7700 2167 cleanup started at [ 400] pbx_realtime.c load_module() 0x7f97fec23700 2165 lock_broker started at [ 524] func_lock.c load_module() 0x7f97fee13700 2161 cal->tech->load_calendar started at [ 489] res_calendar.c build_calendar() 0x7f97fec9f700 2164 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97fed1b700 2163 cal->tech->load_calendar started at [ 489] res_calendar.c build_calendar() 0x7f97fed97700 2162 cal->tech->load_calendar started at [ 489] res_calendar.c build_calendar() 0x7f97fee8f700 2160 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97fef0b700 2159 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97fef87700 2158 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97ff003700 2157 do_monitor started at [11645] chan_dahdi.c restart_monitor() 0x7f97ff07f700 2156 do_monitor started at [29518] chan_sip.c restart_monitor() 0x7f97ff1f3700 2153 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff2eb700 2151 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff26f700 2152 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff3e3700 2149 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff45f700 2148 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff4db700 2147 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff5d3
[asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload'
Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip reload'. This is what I have on CLI : sip5*CLI> sip reload [Oct 7 23:58:40] Reloading SIP [Oct 7 23:58:40] == Parsing '/etc/asterisk/sip.conf': Found [Oct 7 23:58:40] == Parsing '/etc/asterisk/sipTemplates.conf': Found [Oct 7 23:58:40] == Parsing '/etc/asterisk/users.conf': Found [Oct 7 23:58:40] == Using SIP TOS bits 96 [Oct 7 23:58:40] == Using SIP CoS mark 3 [Oct 7 23:58:40] == TLS/SSL ECDH initialized (secp256r1), faster PFS cipher-suites enabled [Oct 7 23:58:40] == TLS/SSL certificate ok --> no more output on CLI. Asterisk has gone completely ! Another 'sip reload' gives : sip5*CLI> sip reload [Oct 8 00:01:10] Previous SIP reload not yet done sip5*CLI> sip reload sip5*CLI> Other commands are no problem on the CLI (while the freeze occurs ! ) : sip5*CLI> core show version Asterisk certified/13.8-cert3 built by root @ sip5.mydomain.tld on a x86_64 running Linux on 2016-10-07 21:27:15 UTC sip5*CLI> sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 0 active SIP channels sip5*CLI> core show threads 0x7f97ff0fb700 2849 netconsole started at [ 1639] asterisk.c listener() 0x7f97fe843700 2760 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff367700 2759 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97fe8bf700 2758 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97fe93b700 2173 monitor_sig_flagsstarted at [ 4768] asterisk.c asterisk_daemon() 0x7f97fe9b7700 2172 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97fea33700 2171 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97feaaf700 2170 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97feb2b700 2169 scan_thread started at [ 920] pbx_spool.c load_module() 0x7f97feba7700 2167 cleanup started at [ 400] pbx_realtime.c load_module() 0x7f97fec23700 2165 lock_broker started at [ 524] func_lock.c load_module() 0x7f97fee13700 2161 cal->tech->load_calendar started at [ 489] res_calendar.c build_calendar() 0x7f97fec9f700 2164 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97fed1b700 2163 cal->tech->load_calendar started at [ 489] res_calendar.c build_calendar() 0x7f97fed97700 2162 cal->tech->load_calendar started at [ 489] res_calendar.c build_calendar() 0x7f97fee8f700 2160 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97fef0b700 2159 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97fef87700 2158 default_tps_processing_function started at [ 200] taskprocessor.c default_listener_start() 0x7f97ff003700 2157 do_monitor started at [11645] chan_dahdi.c restart_monitor() 0x7f97ff07f700 2156 do_monitor started at [29518] chan_sip.c restart_monitor() 0x7f97ff1f3700 2153 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff2eb700 2151 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff26f700 2152 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff3e3700 2149 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff45f700 2148 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff4db700 2147 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff5d3700 2145 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff557700 2146 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff64f700 2144 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff6cb700 2143 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff747700 2142 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff7c3700 2141 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff83f700 2140 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff8bb700 2139 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff937700 2138 worker_start started at [ 1077] threadpool.c worker_thread_start() 0x7f97ff9b3700 2137 worker_start starte
Re: [asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily
On 02-09-16 11:51, Administrator TOOTAI wrote: Le 02/09/2016 à 11:26, Jonas Kellens a écrit : Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal' (thanks to SIP/myaccount184-3729) Question : how can I read the variable which contains the value 'myaccount184' in the context from-internal ? From SIP_HEADER(TO) ? [...] Hello SIP_HEADER(TO) is empty. So is SIP_HEADER(FROM). My dialplan : exten => _ZXX,n,NoOp(CallerIDnum = ${CALLERID(num)} CallerIDall = ${CALLERID(all)}) exten => _ZXX,n,NoOp(sipheaderto = ${SIP_HEADER(TO)}) exten => _ZXX,n,NoOp(sipheaderfrom = ${SIP_HEADER(FROM)}) On the Asterisk CLI : [Sep 22 09:43:04] -- Called SIP/nnsa135 [Sep 22 09:43:04] -- Got SIP response 302 "Moved Temporarily" back from 8.9.10.11:65466 [Sep 22 09:43:04] -- Now forwarding SIP/Incoming-0bd9 to 'Local/208@from-context' (thanks to SIP/nnsa135-0be1) ... [Sep 22 09:43:04] -- Executing [208@from-context:5] NoOp("Local/208@from-context-0079;2", "CallerIDnum = 09210 CallerIDall = "Cpss" <09210>") in new stack [Sep 22 09:43:04] -- Executing [208@from-context:6] NoOp("Local/208@from-context-0079;2", "sipheaderto = ") in new stack [Sep 22 09:43:04] -- Executing [208@from-context:7] NoOp("Local/208@from-context-0079;2", "sipheaderfrom = ") in new stack Any more ideas on how to get the value "nnsa135" (being the SIP username) please ? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ast 13.11.2 : bridgepeer variable empty ?
Hello I can confirm that the variable DIALEDPEERNAME contains the information that I would expect in the variable BRIDGEPEER. But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of Asterisk version 13 ?! So if this is not the intention, then yes this is probably a bug and should be reported. Kind regards. Jonas. On 18-09-16 19:58, Ludovic Gasc wrote: Hi, You might use DIALEDPEERNAME instead of BRIDGEPEER. Nevertheless, I've the same issue with another BRIDGE prefix variable: I never retrieve at one moment BRIDGEPVTCALLID variable, even if it's documented in Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables Nevertheless, the variable seems to be set in the Asterisk source code: https://github.com/asterisk/asterisk/blob/13.10/main/bridge.c#L1222 I see no issues open about that, do I need to open an issue ? Have a nice week. -- Ludovic Gasc (GMLudo) http://www.gmludo.eu/ 2016-09-17 11:47 GMT+02:00 Jonas Kellens <mailto:jonas.kell...@telenet.be>>: Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-0051] app_dial.c: SIP/myprovider-010b is making progress passing it to SIP/mysippeer-0108 [Sep 17 11:30:05] VERBOSE[23420][C-0051] app_dial.c: SIP/myprovider-010b answered SIP/mysippeer-0108 [Sep 17 11:30:05] VERBOSE[23522][C-0051] bridge_channel.c: Channel SIP/myprovider-010b joined 'simple_bridge' basic-bridge [Sep 17 11:30:05] VERBOSE[23420][C-0051] bridge_channel.c: Channel SIP/mysippeer-0108 joined 'simple_bridge' basic-bridge Call ends : [Sep 17 11:34:36] VERBOSE[23420][C-0051] bridge_channel.c: Channel SIP/mysippeer-0108 left 'simple_bridge' basic-bridge [Sep 17 11:34:36] VERBOSE[23522][C-0051] bridge_channel.c: Channel SIP/myprovider-010b left 'simple_bridge' basic-bridge When the call ends in Asterisk version 1.8.32.3 I can read the variable in h-context. In Asterisk 13.11.2 this variable is always empty. How come ?? Dialplan logic : ... exten => h,n,NoOp(bridgepeer = ${BRIDGEPEER}) ... CLI on Asterisk 13.11.2 : -- Executing [h@calling:15] NoOp("SIP/mysippeer-4c80", "bridgepeer = SIP/myprovider-4c83") in new stack CLI on Asterisk 13.11.2 : VERBOSE[23420][C-0051] pbx.c: Executing [h@calling:15] NoOp("SIP/mysippeer-0108", "bridgepeer = ") in new stack What has changed and how can I get the 13.11 version of ${BRIDGEPEER} ?? Thanks in advance ! Kind regards. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference <http://www.asterisk.org/community/astricon-user-conference> New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started <https://wiki.asterisk.org/wiki/display/AST/Getting+Started> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ast 13.11.2 : bridgepeer variable empty ?
Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-0051] app_dial.c: SIP/myprovider-010b is making progress passing it to SIP/mysippeer-0108 [Sep 17 11:30:05] VERBOSE[23420][C-0051] app_dial.c: SIP/myprovider-010b answered SIP/mysippeer-0108 [Sep 17 11:30:05] VERBOSE[23522][C-0051] bridge_channel.c: Channel SIP/myprovider-010b joined 'simple_bridge' basic-bridge [Sep 17 11:30:05] VERBOSE[23420][C-0051] bridge_channel.c: Channel SIP/mysippeer-0108 joined 'simple_bridge' basic-bridge Call ends : [Sep 17 11:34:36] VERBOSE[23420][C-0051] bridge_channel.c: Channel SIP/mysippeer-0108 left 'simple_bridge' basic-bridge [Sep 17 11:34:36] VERBOSE[23522][C-0051] bridge_channel.c: Channel SIP/myprovider-010b left 'simple_bridge' basic-bridge When the call ends in Asterisk version 1.8.32.3 I can read the variable in h-context. In Asterisk 13.11.2 this variable is always empty. How come ?? Dialplan logic : ... exten => h,n,NoOp(bridgepeer = ${BRIDGEPEER}) ... CLI on Asterisk 13.11.2 : -- Executing [h@calling:15] NoOp("SIP/mysippeer-4c80", "bridgepeer = SIP/myprovider-4c83") in new stack CLI on Asterisk 13.11.2 : VERBOSE[23420][C-0051] pbx.c: Executing [h@calling:15] NoOp("SIP/mysippeer-0108", "bridgepeer = ") in new stack What has changed and how can I get the 13.11 version of ${BRIDGEPEER} ?? Thanks in advance ! Kind regards. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue show : failed to extend from 240 to 327
On 10-09-16 09:42, Jonas Kellens wrote: On 10-09-16 00:50, Richard Mudgett wrote: On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello when I type on the Asterisk CLi 'queue show', I first get a list of my queues and then the following : failed to extend from 240 to 327 failed to extend from 240 to 334 I could not find any information on this on the web, except this : https://issues.asterisk.org/jira/browse/ASTERISK-8828 <https://issues.asterisk.org/jira/browse/ASTERISK-8828> which is an old 'bug' that should have been fixed meanwhile. Any more thoughts on why I should be getting this message when asking information about queues (I don't see this message on any other command). That message is a result of trying to build a string where the buffer is too small to contain it. I would expect that there is a truncated string in the 'queue show' output. You haven't stated which Asterisk version you are using. It may already be fixed. Hello I have this with asterisk-certified-13.8-cert1 and also with asterisk-certified-13.8-cert2 Could it be that the membername value (and interface value) in my realtime MySQL table queue_members is too long ?? It looks like this : Local/01_vlaebidvxcrxrheebdin354@ExternalCallFromQueue Local/02_vlaebidvxcrxrheebdin114@ExternalCallFromQueue Local/03_vlaebidvxcrxrheebdin329@ExternalCallFromQueue I have the idea that this is the "problem". FYI : it also makes that Asterisk restarts (with core dump) whenever a queue is addressed. Very /annoying/ ! So string size too large and buffer too small. FYI : I do not have this with any version of Asterisk 1.8. This is a "problem" that exists only in Asterisk 13. How to fix this ?? This is an example output for queue show <> on Asterisk version asterisk-certified-13.8-cert2 (same on asterisk-certified-13.8-cert1) : sip*CLI> queue show cvikbubohirndceiaetsq cvikbubohirndceiaetsq has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: cvikbubohirndceiaets012 (Local/cvikbubohirndceiaets012@ExternalCallFromQueue from Local/cvikbubohirndceiaets012@ExternalCallFromQueue) (ringinuse disabled) (realtime) (Not in use) has taken no cvikbubohirndceiaets248 (Local/cvikbubohirndceiaets248@ExternalCallFromQueue from Local/cvikbubohirndceiaets248@ExternalCallFromQueue) (ringinuse disabled) (realtime) (Not in use) has taken no cvikbubohirndceiaets428 (Local/cvikbubohirndceiaets428@ExternalCallFromQueue from Local/cvikbubohirndceiaets428@ExternalCallFromQueue) (ringinuse disabled) (realtime) (Not in use) has taken no cvikbubohirndceiaets461 (Local/cvikbubohirndceiaets461@ExternalCallFromQueue from Local/cvikbubohirndceiaets461@ExternalCallFromQueue) (ringinuse disabled) (realtime) (Not in use) has taken no cvikbubohirndceiaets629 (Local/cvikbubohirndceiaets629@ExternalCallFromQueue from Local/cvikbubohirndceiaets629@ExternalCallFromQueue) (ringinuse disabled) (realtime) (Not in use) has taken no No Callers failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 Any idea on how to fix this ?? Kind regards. J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue show : failed to extend from 240 to 327
On 10-09-16 00:50, Richard Mudgett wrote: On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello when I type on the Asterisk CLi 'queue show', I first get a list of my queues and then the following : failed to extend from 240 to 327 failed to extend from 240 to 334 I could not find any information on this on the web, except this : https://issues.asterisk.org/jira/browse/ASTERISK-8828 <https://issues.asterisk.org/jira/browse/ASTERISK-8828> which is an old 'bug' that should have been fixed meanwhile. Any more thoughts on why I should be getting this message when asking information about queues (I don't see this message on any other command). That message is a result of trying to build a string where the buffer is too small to contain it. I would expect that there is a truncated string in the 'queue show' output. You haven't stated which Asterisk version you are using. It may already be fixed. Hello I have this with asterisk-certified-13.8-cert1 and also with asterisk-certified-13.8-cert2 Could it be that the membername value (and interface value) in my realtime MySQL table queue_members is too long ?? It looks like this : Local/01_vlaebidvxcrxrheebdin354@ExternalCallFromQueue Local/02_vlaebidvxcrxrheebdin114@ExternalCallFromQueue Local/03_vlaebidvxcrxrheebdin329@ExternalCallFromQueue I have the idea that this is the "problem". FYI : it also makes that Asterisk restarts (with core dump) whenever a queue is addressed. Very /annoying/ ! So string size too large and buffer too small. FYI : I do not have this with any version of Asterisk 1.8. This is a "problem" that exists only in Asterisk 13. How to fix this ?? Kind regards. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue show : failed to extend from 240 to 327
Hello when I type on the Asterisk CLi 'queue show', I first get a list of my queues and then the following : failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 323 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 323 failed to extend from 240 to 323 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 323 failed to extend from 240 to 334 failed to extend from 240 to 334 failed to extend from 240 to 334 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 334 failed to extend from 240 to 334 I could not find any information on this on the web, except this : https://issues.asterisk.org/jira/browse/ASTERISK-8828 which is an old 'bug' that should have been fixed meanwhile. Any more thoughts on why I should be getting this message when asking information about queues (I don't see this message on any other command). Kind regards Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily
Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal' (thanks to SIP/myaccount184-3729) Question : how can I read the variable which contains the value 'myaccount184' in the context from-internal ? The following variables I've tried are empty : ChannelPeerip=${CHANNEL(peerip)} Channelrecvip=${CHANNEL(recvip)} Channelfrom=${CHANNEL(from)} Channeluri=${CHANNEL(uri)} Channeluseragent=${CHANNEL(useragent)}) You can see this on the CLI output here : [Aug 31 14:59:34] -- Executing [23@from-internal:7] NoOp("Local/23@from-internal-07f5;2", "ChannelPeerip= Channelrecvip= Channelfrom=") in new stack [Aug 31 14:59:34] -- Executing [23@from-internal:8] NoOp("Local/23@from-internal-07f5;2", "Channeluri= Channeluseragent=") in new stack Anyone knows the correct variable to read ? Kind regards Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 17-08-16 23:24, George Joseph wrote: On Wed, Aug 17, 2016 at 1:40 PM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: On 16-08-16 17:45, George Joseph wrote: On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: On 16-08-16 04:38, George Joseph wrote: On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello using pjproject 2.5.5 using asterisk-certified-13.8-cert1 IIRC there were API changes in pjproject 2.5 that aren't accounted for in asterisk 13.8. Try pjproject 2.4.5 first and let's see if that works Compiled pjproject 2.5.5 with : ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr Compiled Asterisk 13 with ./configure --libdir=/usr/lib64 All pjproject modules are selectable in menuselect, so here no problem. Modules are present in /usr/lib64/asterisk/module (see below). But when I start asterisk, I get a lot of errors concerning res_pjsip (see below) on the asterisk CLI. Anyone have some input on this ? Thanks. Kind regards. -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com <http://www.digium.com/> & www.asterisk.org <http://www.asterisk.org/> Hello how can I disable all modules related to pjsip in modules.conf ?? I have now : [modules] autoload=yes preload => res_config_mysql.so noload => pbx_gtkconsole.so noload => res_pjsip.so noload => res_pjsip_pubsub.so noload => res_pjsip_session.so noload => chan_pjsip.so noload => res_pjsip_exten_state.so noload => res_pjsip_log_forwarder.so load => res_musiconhold.so noload => chan_alsa.so noload => chan_oss.so noload => chan_console.so This does not make the CLI erros go away. I still have the idea that pjsip is loaded. I'm not sure what your objective is. If you want to completely disable pjsip, run ./configure --without-pjproject. When I compile "--without-pjproject" I loose all webrtc functionality. I get errors about the lack of "ice-frag and ice-pwd in the SDP-body". So I guess I DO need pjproject. But I do not want to use pjsip (I prefer sip). Do you have any other input or idea ? Ok, I get it now. Use pjproject-2.4.5 and in menuselect, disable all the res_pjsip modules. I can confirm that compiling pjproject 2.4.5 (but ALSO pjproject 2.5.5) with asterisk-certified-13.8-cert1 AND "disable all the res_pjsip modules" works fine for me. Kind regards J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 17-08-16 23:17, Jonathan H wrote: On 17 August 2016 at 20:40, Jonas Kellens wrote: When I compile "--without-pjproject" I loose all webrtc functionality. I get errors about the lack of "ice-frag and ice-pwd in the SDP-body". So I guess I DO need pjproject. But I do not want to use pjsip (I prefer sip). Do you have any other input or idea ? Yes. I've never had a problem compiling or installing Asterisk; I simply download the latest version, follow the instructions, and 10 minutes later I'm compiled and up and running. No messing about with weird seperate downloads of unsupported versions of pjsip - I just use the bundled pjsip install and off I go. But from your posts, it seems you want to do modern web stuff like WebRTC and so on, on old version of centos, old versions of asterisk, old version of the SIP channel driver. What particular reason is there to even bother with the certified version - the instructions say the regular most recent LTS download should be first choice. And why do you prefer SIP? pjsip was introduced in Asterisk 12 nearly 3 years ago, and SIP is pretty much deprecated now. As a newbie, I looked at SIP and it all seemed a bit bonkers - "type=friend, insecure=very" - what's THAT all about?! In pjsip, I just setup a pjsip_wizard and template my endpoints in pjsip.conf, and I'm done in a few lines. https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-13-on-Ubuntu.md This is me, creating a brand new Asterisk install on a low end $5 VPS which handles more concurrent calls than I need it to (at least 20 so far!); https://www.youtube.com/watch?v=h12NkJQwpYo (I just found out that the Youtube annotations don't work on mobile, so watch on desktop for it to make sense!). I'm probably the newbiest of noobs here, but just using the latest current stable version of everything available and following the install page on the Asterisk Wiki I can fire up a VPS and be receiving calls in 20 minutes, from scratch. And I'm genuinely interested in why people struggle on for days with old versions of things. I'm not asking all this to create argument, but I am genuinely interested. Perhaps I am missing a major point here? Because in some environments stability is far more important than 'latest' and 'newest'. Kind regards. J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 16-08-16 17:45, George Joseph wrote: On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: On 16-08-16 04:38, George Joseph wrote: On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello using pjproject 2.5.5 using asterisk-certified-13.8-cert1 IIRC there were API changes in pjproject 2.5 that aren't accounted for in asterisk 13.8. Try pjproject 2.4.5 first and let's see if that works Compiled pjproject 2.5.5 with : ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr Compiled Asterisk 13 with ./configure --libdir=/usr/lib64 All pjproject modules are selectable in menuselect, so here no problem. Modules are present in /usr/lib64/asterisk/module (see below). But when I start asterisk, I get a lot of errors concerning res_pjsip (see below) on the asterisk CLI. Anyone have some input on this ? Thanks. Kind regards. -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com <http://www.digium.com/> & www.asterisk.org <http://www.asterisk.org/> Hello how can I disable all modules related to pjsip in modules.conf ?? I have now : [modules] autoload=yes preload => res_config_mysql.so noload => pbx_gtkconsole.so noload => res_pjsip.so noload => res_pjsip_pubsub.so noload => res_pjsip_session.so noload => chan_pjsip.so noload => res_pjsip_exten_state.so noload => res_pjsip_log_forwarder.so load => res_musiconhold.so noload => chan_alsa.so noload => chan_oss.so noload => chan_console.so This does not make the CLI erros go away. I still have the idea that pjsip is loaded. I'm not sure what your objective is. If you want to completely disable pjsip, run ./configure --without-pjproject. When I compile "--without-pjproject" I loose all webrtc functionality. I get errors about the lack of "ice-frag and ice-pwd in the SDP-body". So I guess I DO need pjproject. But I do not want to use pjsip (I prefer sip). Do you have any other input or idea ? Kind regards. J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13
Remove yourself ! Don't hijack my thread ! On 17-08-16 14:53, Dario Estupinan wrote: REMOVE ME please. 2016-08-15 15:16 GMT-05:00 Jonas Kellens <mailto:jonas.kell...@telenet.be>>: Hello after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in MySQL DB) register anymore. [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from 'mailto:sip%3Atestacc77@178.19.90.240>>' failed for '78.119.140.190:5076 <http://78.119.140.190:5076>' - Wrong password [Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from 'mailto:sip%3Atestacc78@178.19.90.240>>' failed for '78.119.140.190:5072 <http://78.119.140.190:5072>' - Wrong password [Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from 'mailto:sip%3Atestacc79@178.19.90.240>>' failed for '78.119.140.190:5062 <http://78.119.140.190:5062>' - Wrong password [Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from 'mailto:sip%3Atestacc80@178.19.90.240>>' failed for '78.119.140.190:5060 <http://78.119.140.190:5060>' - Wrong password [Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from 'mailto:sip%3Atestacc81@178.19.90.240>>' failed for '78.119.140.190:5060 <http://78.119.140.190:5060>' - Wrong password Is this a known problem ?? Second question I have : can I get the complete list of columns that can be used in realtime database for sip peers somewhere (update for Ast 13) ? Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile, dtlssetup possible ?? Thanks for the help. Kind regards. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- ** *DARIO ESTUPIÑAN V.* *Líder de NOC+* *Cel: 3008832295* *E-Mail: darioestupi...@soygenial.co <mailto:darioestupi...@soygenial.co>* Antes de imprimir este mensaje, asegúrese de que es necesario. Proteger el medio ambiente está también en sus manos. AVISO LEGAL: Este mensaje es confidencial, puede contener información privilegiada y no puede ser usado ni divulgado por personas distintas de su destinatario. Si recibe este correo por error, por favor elimínelo y avise a su remitente. Está prohibida su retención, grabación, utilización, aprovechamiento o divulgación con cualquier propósito. La Corporación Politécnica Nacional de Colombia no asume ninguna responsabilidad por eventuales daños generados por el recibo y el uso de este material, siendo responsabilidad del destinatario verificar con sus propios medios la existencia de virus u otros defectos. El presente correo electrónico solo refleja la opinión de su Remitente y no representa necesariamente la opinión oficial de la Corporación o de sus Directivos. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13
On 15-08-16 23:00, Carlos Chavez wrote: I highly recommend that you use alembic to set up your database as this will make sure you are always using the correct database schema. You should be able to find the "official" structure in the contrib/realtime/mysql directory of the Asterisk source. Hello in contrib/realtime/mysql I see a table 'sippeers' with a column "transport ENUM('udp','tcp','tls','ws','wss','udp,tcp','tcp,udp') " but I see no columns dtlsenable, dtlsverify, dtlscertfile, dtlscafile, dtlssetup ? So if we can define a sip peer with transport 'ws' or 'wss', then why are there no columns for the 'dtls'-part (which is kinda mandatory) ? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 16-08-16 04:38, George Joseph wrote: On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello using pjproject 2.5.5 using asterisk-certified-13.8-cert1 IIRC there were API changes in pjproject 2.5 that aren't accounted for in asterisk 13.8. Try pjproject 2.4.5 first and let's see if that works Compiled pjproject 2.5.5 with : ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr Compiled Asterisk 13 with ./configure --libdir=/usr/lib64 All pjproject modules are selectable in menuselect, so here no problem. Modules are present in /usr/lib64/asterisk/module (see below). But when I start asterisk, I get a lot of errors concerning res_pjsip (see below) on the asterisk CLI. Anyone have some input on this ? Thanks. Kind regards. -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com <http://www.digium.com/> & www.asterisk.org <http://www.asterisk.org/> Hello how can I disable all modules related to pjsip in modules.conf ?? I have now : [modules] autoload=yes preload => res_config_mysql.so noload => pbx_gtkconsole.so noload => res_pjsip.so noload => res_pjsip_pubsub.so noload => res_pjsip_session.so noload => chan_pjsip.so noload => res_pjsip_exten_state.so noload => res_pjsip_log_forwarder.so load => res_musiconhold.so noload => chan_alsa.so noload => chan_oss.so noload => chan_console.so This does not make the CLI erros go away. I still have the idea that pjsip is loaded. Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13
Hello after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in MySQL DB) register anymore. [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '' failed for '78.119.140.190:5076' - Wrong password [Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '' failed for '78.119.140.190:5072' - Wrong password [Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '' failed for '78.119.140.190:5062' - Wrong password [Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '' failed for '78.119.140.190:5060' - Wrong password [Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '' failed for '78.119.140.190:5060' - Wrong password Is this a known problem ?? Second question I have : can I get the complete list of columns that can be used in realtime database for sip peers somewhere (update for Ast 13) ? Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile, dtlssetup possible ?? Thanks for the help. Kind regards. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
Hello using pjproject 2.5.5 using asterisk-certified-13.8-cert1 Compiled pjproject 2.5.5 with : ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr Compiled Asterisk 13 with ./configure --libdir=/usr/lib64 All pjproject modules are selectable in menuselect, so here no problem. Modules are present in /usr/lib64/asterisk/module (see below). But when I start asterisk, I get a lot of errors concerning res_pjsip (see below) on the asterisk CLI. Anyone have some input on this ? Thanks. Kind regards. [root@sip admin]# ls /usr/lib64/asterisk/modules | grep pjsip chan_pjsip.so func_pjsip_aor.so func_pjsip_contact.so func_pjsip_endpoint.so res_pjsip_acl.so res_pjsip_authenticator_digest.so res_pjsip_caller_id.so res_pjsip_config_wizard.so res_pjsip_dialog_info_body_generator.so res_pjsip_diversion.so res_pjsip_dlg_options.so res_pjsip_dtmf_info.so res_pjsip_endpoint_identifier_anonymous.so res_pjsip_endpoint_identifier_ip.so res_pjsip_endpoint_identifier_user.so res_pjsip_exten_state.so res_pjsip_header_funcs.so res_pjsip_logger.so res_pjsip_messaging.so res_pjsip_multihomed.so res_pjsip_mwi_body_generator.so res_pjsip_mwi.so res_pjsip_nat.so res_pjsip_notify.so res_pjsip_one_touch_record_info.so res_pjsip_outbound_authenticator_digest.so res_pjsip_outbound_publish.so res_pjsip_outbound_registration.so res_pjsip_path.so res_pjsip_pidf_body_generator.so res_pjsip_pidf_digium_body_supplement.so res_pjsip_pidf_eyebeam_body_supplement.so res_pjsip_publish_asterisk.so res_pjsip_pubsub.so res_pjsip_refer.so res_pjsip_registrar_expire.so res_pjsip_registrar.so res_pjsip_rfc3326.so res_pjsip_sdp_rtp.so res_pjsip_send_to_voicemail.so res_pjsip_session.so res_pjsip_sips_contact.so res_pjsip.so res_pjsip_t38.so res_pjsip_transport_management.so res_pjsip_transport_websocket.so res_pjsip_xpidf_body_generator.so Asterisk CLI : [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error loading module 'res_pjsip_registrar.so': /usr/lib64/asterisk/modules/res_pjsip_registrar.so: undefined symbol: ast_sip_location_retrieve_aor_contacts_nolock [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 'res_pjsip_registrar.so' could not be loaded. [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error loading module 'res_pjsip_path.so': /usr/lib64/asterisk/modules/res_pjsip_path.so: undefined symbol: ast_sip_location_retrieve_aor [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 'res_pjsip_path.so' could not be loaded. [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error loading module 'res_pjsip_authenticator_digest.so': /usr/lib64/asterisk/modules/res_pjsip_authenticator_digest.so: undefined symbol: ast_sip_retrieve_auths [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 'res_pjsip_authenticator_digest.so' could not be loaded. [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error loading module 'res_pjsip_dialog_info_body_generator.so': /usr/lib64/asterisk/modules/res_pjsip_dialog_info_body_generator.so: undefined symbol: ast_sip_pubsub_unregister_body_generator [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 'res_pjsip_dialog_info_body_generator.so' could not be loaded. [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error loading module 'res_pjsip_sdp_rtp.so': /usr/lib64/asterisk/modules/res_pjsip_sdp_rtp.so: undefined symbol: ast_sip_session_unregister_supplement [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 'res_pjsip_sdp_rtp.so' could not be loaded. [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error loading module 'res_pjsip_publish_asterisk.so': /usr/lib64/asterisk/modules/res_pjsip_publish_asterisk.so: undefined symbol: ast_sip_register_publish_handler [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 'res_pjsip_publish_asterisk.so' could not be loaded. [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error loading module 'res_pjsip_send_to_voicemail.so': /usr/lib64/asterisk/modules/res_pjsip_send_to_voicemail.so: undefined symbol: ast_sip_session_unregister_supplement [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 'res_pjsip_send_to_voicemail.so' could not be loaded. [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error loading module 'res_pjsip_diversion.so': /usr/lib64/asterisk/modules/res_pjsip_diversion.so: undefined symbol: ast_sip_session_unregister_supplement [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 'res_pjsip_diversion.so' could not be loaded. [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error loading module 'res_pjsip_dlg_options.so': /usr/lib64/asterisk/modules/res_pjsip_dlg_options.so: undefined symbol: ast_sip_sessio
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello I've succeeded in installing Asterisk 13 and more important : I can make webRTC call and I have audio !! For those on the search like myself, I want to spare some weeks of headache. My steps (CentOS 6.8) : yum install uuid-devel libuuid-devel autoconf patch automake libcurl-devel libogg-devel libvorbis-devel speex-devel popt-devel libtool-ltdl-devel libresample-devel gsm-devel libedit-devel python-devel jansson-devel binutils-devel wget http://www.pjsip.org/release/2.5.5/pjproject-2.5.5.tar.bz2 tar -xjvf pjproject-2.5.5.tar.bz2 ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr make dep make make install ldconfig -p | grep pj ldconfig wget http://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-13.8-current.tar.gz [root@siptest asterisk-certified-13.8-cert1]# ./configure --libdir=/usr/lib64 [root@siptest asterisk-certified-13.8-cert1]# make menuselect [root@siptest asterisk-certified-13.8-cert1]# make && make install Forget the option "--with-pjproject-bundled" I would say. Did not work for me on : CentOS release 6.8 (Final) Kind regards. On 12-08-16 17:22, Jonas Kellens wrote: Hello running into several problems when installing asterisk-certified-13.8-cert1 (more then I ever had in Asterisk 11 and 12). I compile : ./configure --libdir=/usr/lib64 --with-pjproject-bundled First, I do not seem to have res_srtp module available, although all necessary libs are present on the system Second, I am not able to start Asterisk with following error : "/usr/sbin/asterisk: error while loading shared libraries: libpj.so.2: cannot open shared object file: No such file or directory" Help appreciated. Kind regards. On 12-08-16 16:58, Jonas Kellens wrote: On 12-08-16 16:38, Joshua Colp wrote: Jonas Kellens wrote: Question : I noticed I received an error when installing pjproject --with-external-srtp I do not seems to have the srtp capability. (However I can easily install with "yum install libsrtp-devel") Can this have anything to do with the no-audio-problems that I'm having ?? WebRTC requires SRTP and Asterisk has to be built with it enabled. It's okay if pjproject doesn't as we don't use their media layer. Do you have the res_srtp module in Asterisk? Hello Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest version Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version However, I am not able to select res_srtp module in menuselect. It says XXX res_srtp module Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello running into several problems when installing asterisk-certified-13.8-cert1 (more then I ever had in Asterisk 11 and 12). I compile : ./configure --libdir=/usr/lib64 --with-pjproject-bundled First, I do not seem to have res_srtp module available, although all necessary libs are present on the system Second, I am not able to start Asterisk with following error : "/usr/sbin/asterisk: error while loading shared libraries: libpj.so.2: cannot open shared object file: No such file or directory" Help appreciated. Kind regards. On 12-08-16 16:58, Jonas Kellens wrote: On 12-08-16 16:38, Joshua Colp wrote: Jonas Kellens wrote: Question : I noticed I received an error when installing pjproject --with-external-srtp I do not seems to have the srtp capability. (However I can easily install with "yum install libsrtp-devel") Can this have anything to do with the no-audio-problems that I'm having ?? WebRTC requires SRTP and Asterisk has to be built with it enabled. It's okay if pjproject doesn't as we don't use their media layer. Do you have the res_srtp module in Asterisk? Hello Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest version Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version However, I am not able to select res_srtp module in menuselect. It says XXX res_srtp module Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
On 12-08-16 16:38, Joshua Colp wrote: Jonas Kellens wrote: Question : I noticed I received an error when installing pjproject --with-external-srtp I do not seems to have the srtp capability. (However I can easily install with "yum install libsrtp-devel") Can this have anything to do with the no-audio-problems that I'm having ?? WebRTC requires SRTP and Asterisk has to be built with it enabled. It's okay if pjproject doesn't as we don't use their media layer. Do you have the res_srtp module in Asterisk? Hello Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest version Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version However, I am not able to select res_srtp module in menuselect. It says XXX res_srtp module Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Question : I noticed I received an error when installing pjproject --with-external-srtp I do not seems to have the srtp capability. (However I can easily install with "yum install libsrtp-devel") Can this have anything to do with the no-audio-problems that I'm having ?? Kind regards. On 12-08-16 15:02, Jonas Kellens wrote: Hello setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio. Why do you think this is a NAT issue ? IP and port information in SDP-body is correct. Kind regards. On 12-08-16 09:25, Антон Сацкий wrote: Try delete nat from 77wrtc settings ice should do the same On Aug 11, 2016 10:00 PM, "Jonas Kellens" <mailto:jonas.kell...@telenet.be>> wrote: On 11-08-16 18:03, Matt Fredrickson wrote: On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache :-) I will do so if there is no other option. But still, I don't see why Ast 13 would differ so much in this case ? If ICE and NAT is working (not causing problems) why should Ast 13 bring me audio and Ast 12 don't ?? If you want to minimize grief, start with 13 - WebRTC has been a moving target for the last 5 years, it is not an old, mature standard like ISDN or SIP. If you find interop problems in an older version of Asterisk with WebRTC, it's likely that it has been fixed in 13, and if it hasn't the most likely place to obtain the fix will be in 13. After you get the WebRTC part working, then you can move back the versions of Asterisk you're using to see if it still works. As far as ICE not working goes, if the browser you're talking to is not on the same network as the Asterisk server, it's *possible* you might need a true TURN server as well, instead of just an ICE server. Matthew Fredrickson Matthew when I set the following in rtp.conf : turnaddr=192.158.29.39:3478?transport=udp <http://192.158.29.39:3478?transport=udp> turnusername=28224511:1379330808 turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA then Asterisk 12 gets really slow and sometimes unresponsive. Calls result in 480 request timeout (possibly due to the freeze of Asterisk). So this is also no solution. Can not even test if it brings me some audio in my webRTC calls. (putting the above lines back in comment resolves the issue of Asterisk freeze. This is all EXTREMELY BUGGY !) Asterisk 13 here I come (with very high expectations). Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio. Why do you think this is a NAT issue ? IP and port information in SDP-body is correct. Kind regards. On 12-08-16 09:25, Антон Сацкий wrote: Try delete nat from 77wrtc settings ice should do the same On Aug 11, 2016 10:00 PM, "Jonas Kellens" <mailto:jonas.kell...@telenet.be>> wrote: On 11-08-16 18:03, Matt Fredrickson wrote: On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache :-) I will do so if there is no other option. But still, I don't see why Ast 13 would differ so much in this case ? If ICE and NAT is working (not causing problems) why should Ast 13 bring me audio and Ast 12 don't ?? If you want to minimize grief, start with 13 - WebRTC has been a moving target for the last 5 years, it is not an old, mature standard like ISDN or SIP. If you find interop problems in an older version of Asterisk with WebRTC, it's likely that it has been fixed in 13, and if it hasn't the most likely place to obtain the fix will be in 13. After you get the WebRTC part working, then you can move back the versions of Asterisk you're using to see if it still works. As far as ICE not working goes, if the browser you're talking to is not on the same network as the Asterisk server, it's *possible* you might need a true TURN server as well, instead of just an ICE server. Matthew Fredrickson Matthew when I set the following in rtp.conf : turnaddr=192.158.29.39:3478?transport=udp <http://192.158.29.39:3478?transport=udp> turnusername=28224511:1379330808 turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA then Asterisk 12 gets really slow and sometimes unresponsive. Calls result in 480 request timeout (possibly due to the freeze of Asterisk). So this is also no solution. Can not even test if it brings me some audio in my webRTC calls. (putting the above lines back in comment resolves the issue of Asterisk freeze. This is all EXTREMELY BUGGY !) Asterisk 13 here I come (with very high expectations). Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
On 11-08-16 18:03, Matt Fredrickson wrote: On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens wrote: My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache :-) I will do so if there is no other option. But still, I don't see why Ast 13 would differ so much in this case ? If ICE and NAT is working (not causing problems) why should Ast 13 bring me audio and Ast 12 don't ?? If you want to minimize grief, start with 13 - WebRTC has been a moving target for the last 5 years, it is not an old, mature standard like ISDN or SIP. If you find interop problems in an older version of Asterisk with WebRTC, it's likely that it has been fixed in 13, and if it hasn't the most likely place to obtain the fix will be in 13. After you get the WebRTC part working, then you can move back the versions of Asterisk you're using to see if it still works. As far as ICE not working goes, if the browser you're talking to is not on the same network as the Asterisk server, it's *possible* you might need a true TURN server as well, instead of just an ICE server. Matthew Fredrickson Matthew when I set the following in rtp.conf : turnaddr=192.158.29.39:3478?transport=udp turnusername=28224511:1379330808 turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA then Asterisk 12 gets really slow and sometimes unresponsive. Calls result in 480 request timeout (possibly due to the freeze of Asterisk). So this is also no solution. Can not even test if it brings me some audio in my webRTC calls. (putting the above lines back in comment resolves the issue of Asterisk freeze. This is all EXTREMELY BUGGY !) Asterisk 13 here I come (with very high expectations). Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
11 16:01:20] Got RTP packet from178.119.146.190:58814 (type 08, seq 014125, ts 3292376087, len 000160) [Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via ICE) (type 08, seq 033798, ts 3292376080, len 000160) [Aug 11 16:01:20] Got RTP packet from178.119.146.190:58814 (type 08, seq 014126, ts 3292376247, len 000160) [Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via ICE) (type 08, seq 033799, ts 3292376240, len 000160) [Aug 11 16:01:20] Got RTP packet from178.119.146.190:58814 (type 08, seq 014127, ts 3292376407, len 000160) [Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via ICE) (type 08, seq 033800, ts 3292376400, len 000160) [Aug 11 16:01:20] Got RTP packet from178.119.146.190:58814 (type 08, seq 014128, ts 3292376567, len 000160) On 10-08-16 22:03, Matt Fredrickson wrote: My suggestion is to verify and debug against Asterisk 13 first, and then you can try backing down versions, rather than reverse. WebRTC is a rapidly moving target, and has required ongoing changes that may not have made it into older and feature frozen versions of Asterisk. Matthew Fredrickson On Wed, Aug 10, 2016 at 3:01 PM, Jonas Kellens wrote: Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: I don't see an ice-ufrag or ice-pwd line in the response from Asterisk, correlating with your suspicion that there is no ICE. Are you sure that the stun server you're using (the google one) still works? I haven't tried that server in a while, but I distantly seem to recall that maybe they shut it down. Asterisk 13 is a better place to be as well. Asterisk 11 hasn't been feature updated in a while, and it could be that it could be a number of patches/fixes behind with regards to webrtc support, particularly with regards to interoperating with a modern browser version. Hope that helps, Matthew Fredrickson On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens wrote: On 10-08-16 08:52, Ludovic Gasc wrote: For WebRTC, I recommend you to use Asterisk 13+. Have a nice day. Ludovic Gasc (GMLudo) http://www.gmludo.eu/ Hello then why is there an option in sip.conf and rtp.conf " icesupport=yes" ?? This is no answer to my question. So again : what am I missing to get ICE support on my Asterisk 11.23.0 ?? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache :-) I will do so if there is no other option. But still, I don't see why Ast 13 would differ so much in this case ? If ICE and NAT is working (not causing problems) why should Ast 13 bring me audio and Ast 12 don't ?? I indeed use SIPML5 demo as quick test-case. So do many tutorials on the web. Self-signed certificates should be OK as long as they are imported in the browser. Never knew this could cause audio problems ? Kind regards. On 11-08-16 16:25, Jonathan H wrote: I'm genuinely fascinated why you are insisting on using a version of Asterisk almost 3 years old, for which EOL support ended last year. Is there any particular reason you cannot or will not use the current version as others have suggested? Also, I see you are using Doubango and WebRTC, but in the logs, I see WS and WSS. You NEED to be using 100% WSS otherwise you've not got a hope in hell of anything working with WEBRTC. Check the console of the web browser you are trying to make the call from (CTRL-SHIFT-I in Chrome on Windows, for example). Also, you'll need to be using valid certificates - self-signed certificates won't work for any current implementation of WebRTC that I know of, certainly not if anything involves current versions of Chrome or Firefox. That said, LetsEncrypt certs work fine for this, so no need to spend out on one. Switch to Asterisk 13.10 and save yourself a whole lotta headache. On 11 August 2016 at 15:09, Jonas Kellens <mailto:jonas.kell...@telenet.be>> wrote: Hello Using Asterisk 12.8.2. On 10-08-16 22:03, Matt Fredrickson wrote: My suggestion is to verify and debug against Asterisk 13 first, and then you can try backing down versions, rather than reverse. WebRTC is a rapidly moving target, and has required ongoing changes that may not have made it into older and feature frozen versions of Asterisk. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: I don't see an ice-ufrag or ice-pwd line in the response from Asterisk, correlating with your suspicion that there is no ICE. Are you sure that the stun server you're using (the google one) still works? I haven't tried that server in a while, but I distantly seem to recall that maybe they shut it down. Asterisk 13 is a better place to be as well. Asterisk 11 hasn't been feature updated in a while, and it could be that it could be a number of patches/fixes behind with regards to webrtc support, particularly with regards to interoperating with a modern browser version. Hope that helps, Matthew Fredrickson On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens wrote: On 10-08-16 08:52, Ludovic Gasc wrote: For WebRTC, I recommend you to use Asterisk 13+. Have a nice day. Ludovic Gasc (GMLudo) http://www.gmludo.eu/ Hello then why is there an option in sip.conf and rtp.conf " icesupport=yes" ?? This is no answer to my question. So again : what am I missing to get ICE support on my Asterisk 11.23.0 ?? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
On 10-08-16 08:52, Ludovic Gasc wrote: For WebRTC, I recommend you to use Asterisk 13+. Have a nice day. Ludovic Gasc (GMLudo) http://www.gmludo.eu/ Hello then why is there an option in sip.conf and rtp.conf " icesupport=yes" ?? This is no answer to my question. So again : what am I missing to get ICE support on my Asterisk 11.23.0 ?? Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello I'm trying for several days now to get ICE support for my Asterisk 11.23 on CentOS 6. My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230 --> softphone Zoiper (problem : no audio) Reverse does not work either. (problem : failed get local SDP) I followed this guide : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support I researched on the web and found this useful thread : http://forums.digium.com/viewtopic.php?f=1&t=90167 This is no question "what is wrong ?". I know what is wrong : I need ICE support ! So the question here is : how to get ICE support in my Asterisk ? I've compiled asterisk as follow : [root@myserver admin]# yum install uuid-devel libuuid-devel [root@myserver admin]# ./configure --libdir=/usr/lib64 [root@myserver admin]# make menuselect [root@myserver admin]# make && make install In my sip.conf I have : icesupport = yes In my rtp.conf I have : icesupport=yes stunaddr=stun.l.google.com:19302 My SIP peer definition for webRTC client (sipml5) : [77wrtc] type=peer host=dynamic username=77wrtc defaultuser=77wrtc fromuser=77wrtc secret=987654 disallow=all allow=alaw ;allow=gsm qualify=yes canreinvite=no dtmfmode=rfc2833 amaflags=billing context=testwebrtc nat=force_rport,comedia transport=udp,ws,wss encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=no dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlscafile=/etc/asterisk/keys/ca.crt dtlssetup=actpass SIP registration works fine : [Aug 9 22:12:00] == WebSocket connection from '178.119.146.190:36940' for protocol 'sip' accepted using version '13' [Aug 9 22:12:00] -- Registered SIP '77wrtc' at 178.119.146.190:36940 [Aug 9 22:12:00]> Saved useragent "IM-client/OMA1.0 sipML5-v1.2016.03.04" for peer 77wrtc But when I call from my webRTc client (sipml5 website demo) I have no audio. I think this is because there is no ICE support. You can see in de SIP trace below and the RTP trace below that there is no ICE support in Asterisk. [Aug 9 22:15:50] <--- SIP read from WS:178.119.146.190:36940 ---> [Aug 9 22:15:50] INVITE sip:419@178.18.90.230 SIP/2.0 [Aug 9 22:15:50] Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;rport [Aug 9 22:15:50] From: "77";tag=sRCvFQq3gUMqkl6TKTcl [Aug 9 22:15:50] To: [Aug 9 22:15:50] Contact: "77";+g.oma.sip-im;language="en,fr" [Aug 9 22:15:50] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32 [Aug 9 22:15:50] CSeq: 21553 INVITE [Aug 9 22:15:50] Content-Type: application/sdp [Aug 9 22:15:50] Content-Length: 1815 [Aug 9 22:15:50] Max-Forwards: 70 [Aug 9 22:15:50] Authorization: Digest username="77wrtc",realm="178.18.90.230",nonce="1d8fa83d",uri="sip:419@178.18.90.230",response="cd2da8d1cbf0a2795b38b2048a3a3c49",algorithm=MD5 [Aug 9 22:15:50] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 [Aug 9 22:15:50] Organization: Doubango Telecom [Aug 9 22:15:50] [Aug 9 22:15:50] v=0 [Aug 9 22:15:50] o=- 9108976588890881000 2 IN IP4 127.0.0.1 [Aug 9 22:15:50] s=Doubango Telecom - chrome [Aug 9 22:15:50] t=0 0 [Aug 9 22:15:50] a=group:BUNDLE audio [Aug 9 22:15:50] a=msid-semantic: WMS BJSlrOtzPj6wzI3QugifY58Oi18zpEbkNsps [Aug 9 22:15:50] m=audio 41178 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 [Aug 9 22:15:50] c=IN IP4 178.119.146.190 [Aug 9 22:15:50] a=rtcp:42197 IN IP4 178.119.146.190 [Aug 9 22:15:50] a=candidate:1668076467 1 udp 2122260223 192.168.1.122 41178 typ host generation 0 [Aug 9 22:15:50] a=candidate:1668076467 2 udp 2122260222 192.168.1.122 42197 typ host generation 0 [Aug 9 22:15:50] a=candidate:3794064647 1 udp 1686052607 178.119.146.190 41178 typ srflx raddr 192.168.1.122 rport 41178 generation 0 [Aug 9 22:15:50] a=candidate:3794064647 2 udp 1686052606 178.119.146.190 42197 typ srflx raddr 192.168.1.122 rport 42197 generation 0 [Aug 9 22:15:50] a=candidate:770649923 1 tcp 1518280447 192.168.1.122 0 typ host tcptype active generation 0 [Aug 9 22:15:50] a=candidate:770649923 2 tcp 1518280446 192.168.1.122 0 typ host tcptype active generation 0 [Aug 9 22:15:50] a=ice-ufrag:cd8nLIL1irEPdLZt [Aug 9 22:15:50] a=ice-pwd:97awKXGiAt1TO5jlmb3GMXRy [Aug 9 22:15:50] a=fingerprint:sha-256 A2:EF:18:69:AE:9D:D9:90:45:0E:0D:84:5C:A4:AE:59:1C:53:09:11:F2:10:DF:F9:BB:20:E0:9D:6D:ED:BC:13 [Aug 9 22:15:50] a=setup:actpass [Aug 9 22:15:50] a=mid:audio [Aug 9 22:15:50] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level [Aug 9 22:15:50] a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time [Aug 9 22:15:50] a=sendrecv [Aug 9 22:15:50] a=rtcp-mux [Aug 9 22:15:50] a=rtpmap:111 opus/48000/2 [Aug 9 22:15:50] a=fmtp:111 minptime=10; useinbandfec=1 [Aug 9 22:15:50] a=rtpmap:103 ISAC/16000 [Aug 9 22:15:50] a=rtpmap:104 ISAC/32000 [Aug 9 22:15:50] a=rtpmap:9 G722/8000 [Aug 9 22:15:50] a=r
Re: [asterisk-users] Setting realm=blabla in sip.conf ignored ?
Hello nobody who can help me with this realm issue ?? On 21-06-16 16:36, Jonas Kellens wrote: Hello no matter what I set in sip.conf for the param "realm=blablabla" , I notice in a wireshark trace file that the realm is completely ignored. I see that realm value is still 'asterisk', being the default. Why is this ? (I would like to add a printscreen of the wiresharl trace but then this thread is rejected due to message size) So how can I really change the realm value ? Thanks. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting realm=blabla in sip.conf ignored ?
Hello no matter what I set in sip.conf for the param "realm=blablabla" , I notice in a wireshark trace file that the realm is completely ignored. I see that realm value is still 'asterisk', being the default. Why is this ? (I would like to add a printscreen of the wiresharl trace but then this thread is rejected due to message size) So how can I really change the realm value ? Thanks. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] function SHARED and function IMPORT : 2 questions
Hello I am trying to use the functions SHARED and IMPORT to share variables across SIP-channels. During my use I encounter 2 problems/questions. Question 1. only 1 shared variable per channel ?? When I set 2 shared variables on a channel, and I read them a bit futher in the dialplan, there is only 1 variable that has its value : Dialplan : exten => s,n,Set(SHARED(TheINPUT)=1) exten => s,n,Set(SHARED(MyVar)=222) ... exten => s,n,Set(Import=${IMPORT(${CHANNEL},TheINPUT)}) exten => s,n,Set(ImportAdv=${IMPORT(${CHANNEL},MyVar)}) Execution : [Mar 2 14:19:20] -- Executing [s@routing:56] Set("SIP/980419-0016", "SHARED(TheINPUT)=1") in new stack [Mar 2 14:19:20] -- Executing [s@routing:57] Set("SIP/980419-0016", "SHARED(MyVar)=222") in new stack ... [Mar 2 14:19:26] -- Executing [s@routing:80] Set("SIP/980419-0016", "Import=1") in new stack [Mar 2 14:19:26] -- Executing [s@routing:81] Set("SIP/980419-0016", "ImportAdv=") in new stack As you can see, only variable "TheINPUT" has its value ( 1 ). Variable 'MyVar' is empty. How come ?? Question 2 : how to set a variable on another channel ?? I try to set a Shared Variable on 1 channel : exten => s,n,Set(SHARED(TheINPUT,${BRIDGECH})=1) And read the variable on another channel : exten => s,n,Set(Import=${IMPORT(${BRIDGECH},TheINPUT)}) exten => s,n,Set(Hell=${IMPORT(TheINPUT)}) Execution : (here the shared var is set) [Mar 2 14:58:44] -- Executing [s@routing:58] NoOp("SIP/980419-0025", "bridgech = SIP/SipT01-0021") in new stack [Mar 2 14:58:44] -- Executing [s@routing:59] Set("SIP/980419-0025", "SHARED(TheINPUT,SIP/SipT01-0021)=1") in new stack (here the hared var is read) [Mar 2 14:58:54] -- Executing [s@routing:42] Set("SIP/SipT01-0021", "Import=") in new stack [Mar 2 14:58:54] -- Executing [s@routing:43] Set("SIP/SipT01-0021", "Hell=") in new stack So why is the shared variable "TheINPUT" empty on the channel SIP/SipT01-0021 ?? It clealry has been set by the channel SIP/980419-0025. Thank you for answering these 2 questions of mine. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge
Hello so I got this working with Google Calendar and meanwhile also with MS Exchange. Does anyone have a working example with Horde Calendar (kronolith)? This one seems very tough ! Kind regards Jonas. On 27-10-15 14:52, Jonas Kellens wrote: Mark thank you for your input. I am using Asterisk 1.8.32.3 (latest). I indeed use the "Private iCal url" as presented by Google in Calendar settings. This is my calendar.conf : [cal0] type = caldav url = https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics user = i...@domain.tld secret = mysecretpasswd refresh = 15 timeframe = 60 [cal1] type = ical url = https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics user = i...@domain.tld secret = mysecretpasswd refresh = 15 timeframe = 60 [cal2] type = caldav url = https://www.google.com/calendar/dav/i...@domain.tld/events/ ; Main GMail calendar (the trailing slash is significant!) user = i...@domain.tld secret = mysecretpasswd refresh = 15 timeframe = 60 [cal3] type = ical url = https://www.google.com/calendar/dav/i...@domain.tld/events/ ; Main GMail calendar (the trailing slash is significant!) user = i...@domain.tld secret = mysecretpasswd refresh = 15 timeframe = 60 You see that I try every combination possible. [Oct 27 14:28:28] WARNING[26748]: res_calendar_caldav.c:118 auth_credentials: Invalid username or password for CalDAV calendar 'cal2' [Oct 27 14:28:28] WARNING[26748]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal2, request REPORT to /calendar/dav/i...@domain.tld/events/: Could not authenticate to server: rejected Basic challenge [Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:117 auth_credentials: Invalid username or password for iCalendar 'cal3' [Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:150 fetch_icalendar: Unable to retrieve iCalendar 'cal3' from 'https://www.google.com/calendar/dav/i...@domain.tld/events/': Could not authenticate to server: rejected Basic challenge [Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:477 ical_load_calendar: Unable to parse iCalendar 'cal3' Calendar Type Status -- 77cal3 ical free 77cal2 caldav free 77cal1 ical busy 77cal0 caldav free It seems I finally have a working example !! Namely : [cal1] type = ical url = https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics user = i...@domain.tld secret = mysecretpasswd refresh = 15 timeframe = 60 So the "Private iCal url" of Google Calendar is the one to go ! Jonas. On 27-10-15 14:04, Mark Wiater wrote: On 10/27/2015 8:56 AM, Jonas Kellens wrote: I have changed this setting at Google but it brings me no success. Jonas, I've been using google calendar and Asterisk 1.8 for a couple of years now without issue. I have a note in my configuration that says that I'm using the Private ICAL URL from gmail and that it's the only one that worked for me. Is that the URL that you're using? Did you change your type to ical in calendar.conf? Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge
Mark thank you for your input. I am using Asterisk 1.8.32.3 (latest). I indeed use the "Private iCal url" as presented by Google in Calendar settings. This is my calendar.conf : [cal0] type = caldav url = https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics user = i...@domain.tld secret = mysecretpasswd refresh = 15 timeframe = 60 [cal1] type = ical url = https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics user = i...@domain.tld secret = mysecretpasswd refresh = 15 timeframe = 60 [cal2] type = caldav url = https://www.google.com/calendar/dav/i...@domain.tld/events/ ; Main GMail calendar (the trailing slash is significant!) user = i...@domain.tld secret = mysecretpasswd refresh = 15 timeframe = 60 [cal3] type = ical url = https://www.google.com/calendar/dav/i...@domain.tld/events/ ; Main GMail calendar (the trailing slash is significant!) user = i...@domain.tld secret = mysecretpasswd refresh = 15 timeframe = 60 You see that I try every combination possible. [Oct 27 14:28:28] WARNING[26748]: res_calendar_caldav.c:118 auth_credentials: Invalid username or password for CalDAV calendar 'cal2' [Oct 27 14:28:28] WARNING[26748]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal2, request REPORT to /calendar/dav/i...@domain.tld/events/: Could not authenticate to server: rejected Basic challenge [Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:117 auth_credentials: Invalid username or password for iCalendar 'cal3' [Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:150 fetch_icalendar: Unable to retrieve iCalendar 'cal3' from 'https://www.google.com/calendar/dav/i...@domain.tld/events/': Could not authenticate to server: rejected Basic challenge [Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:477 ical_load_calendar: Unable to parse iCalendar 'cal3' Calendar Type Status -- 77cal3 ical free 77cal2 caldav free 77cal1 ical busy 77cal0 caldav free It seems I finally have a working example !! Namely : [cal1] type = ical url = https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics user = i...@domain.tld secret = mysecretpasswd refresh = 15 timeframe = 60 So the "Private iCal url" of Google Calendar is the one to go ! Jonas. On 27-10-15 14:04, Mark Wiater wrote: On 10/27/2015 8:56 AM, Jonas Kellens wrote: I have changed this setting at Google but it brings me no success. Jonas, I've been using google calendar and Asterisk 1.8 for a couple of years now without issue. I have a note in my configuration that says that I'm using the Private ICAL URL from gmail and that it's the only one that worked for me. Is that the URL that you're using? Did you change your type to ical in calendar.conf? Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge
Hello I have changed this setting at Google but it brings me no success. I add that I have the same problem with another calendar (Horde) : [Oct 27 12:08:32] WARNING[24844]: res_calendar_caldav.c:118 auth_credentials: Invalid username or password for CalDAV calendar 'cal2' [Oct 27 12:08:32] WARNING[24844]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal2, request REPORT to /rpc.php/kronolith/jo...@mydomain.tld/jo...@mydomain.tld.ics: Could not authenticate to server: rejected Basic challenge When using "caldav" or "ical" with Google Calendar, I now get this notice : [Oct 27 13:43:51] WARNING[25202]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal0, request REPORT to /calendar/ical/info%40domain.tld/private-6e3543acbc7e2ad02b3d414124a/basic.ics: SSL handshake failed: SSL error: GnuTLS internal error. I have also taken the URL as presented by Google in the Calendar settings. But as u see... it does not work ! So it does not seem to be a problem with Google Calendar, as the same problem occurs with Horde Calendar. Anyone has a working example please ?? Kind regards Jonas. On 27-10-15 13:19, Dan Heywood wrote: Hi Jonas, Is it google apps? Try checking the following in your google account settings: Allow less secure apps: ON Some non-Google apps and devices use less secure sign-in technology, which could leave your account vulnerable. You can turn off access for these apps (which we recommend) or choose to use them despite the risks. I had to enable this to allow login from a linux based application in order to send out email. Thanks, Dan *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, October 27, 2015 1:33 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge Hello I have changed type 'caldav' to 'ical', but still no succes : [Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:117 auth_credentials: Invalid username or password for iCalendar 'cal1' [Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:150 fetch_icalendar: Unable to retrieve iCalendar 'cal1' from 'https://www.google.com/calendar/dav/i...@mydomain.tld/events/': Could not authenticate to server: rejected Basic challenge siptest*CLI> calendar show calendars Calendar Type Status -- cal1ical free Am I missing something obvious here ? Kind regards Jonas On 26-10-15 17:02, Marek Červenka wrote: try ical url caldav switched to Oauth https://blog.mozilla.org/calendar/2013/09/google-is-changing-the-location-url-of-their-caldav-calendars/ and this looks like you must use Oauth 2.0 https://developers.google.com/google-apps/calendar/caldav/v2/guide Dne 26.10.2015 v 12:17 Jonas Kellens napsal(a): Hello I find very little feedback on the following warning/error when trying to connect to Google calendar : [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118 auth_credentials: Invalid username or password for CalDAV calendar 'cal1' [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal1, request REPORT to /calendar/dav/i...@mydomain.tld/events/ <mailto:/calendar/dav/i...@mydomain.tld/events/>: Could not authenticate to server: rejected Basic challenge [cal1] type = caldav url = https://www.google.com/calendar/dav/i...@mydomain.tld/events/ user = i...@mydomain.tld <mailto:i...@mydomain.tld> secret = mysecret refresh = 15 timeframe = 60 When I go to the URL https://www.google.com/calendar/dav/i...@mydomain.tld/events/ I can log in with the credentials to the calendar (and get a download window for the calendar file). So it seems not a problem of authentication to me. But what then could be the real issue here ? Thanks Kind regards Jonas. -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge
Hello I have changed type 'caldav' to 'ical', but still no succes : [Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:117 auth_credentials: Invalid username or password for iCalendar 'cal1' [Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:150 fetch_icalendar: Unable to retrieve iCalendar 'cal1' from 'https://www.google.com/calendar/dav/i...@mydomain.tld/events/': Could not authenticate to server: rejected Basic challenge siptest*CLI> calendar show calendars Calendar Type Status -- cal1ical free Am I missing something obvious here ? Kind regards Jonas On 26-10-15 17:02, Marek Červenka wrote: try ical url caldav switched to Oauth https://blog.mozilla.org/calendar/2013/09/google-is-changing-the-location-url-of-their-caldav-calendars/ and this looks like you must use Oauth 2.0 https://developers.google.com/google-apps/calendar/caldav/v2/guide Dne 26.10.2015 v 12:17 Jonas Kellens napsal(a): Hello I find very little feedback on the following warning/error when trying to connect to Google calendar : [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118 auth_credentials: Invalid username or password for CalDAV calendar 'cal1' [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal1, request REPORT to /calendar/dav/i...@mydomain.tld/events/: Could not authenticate to server: rejected Basic challenge [cal1] type = caldav url = https://www.google.com/calendar/dav/i...@mydomain.tld/events/ user = i...@mydomain.tld secret = mysecret refresh = 15 timeframe = 60 When I go to the URL https://www.google.com/calendar/dav/i...@mydomain.tld/events/ I can log in with the credentials to the calendar (and get a download window for the calendar file). So it seems not a problem of authentication to me. But what then could be the real issue here ? Thanks Kind regards Jonas. -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge
Hello I find very little feedback on the following warning/error when trying to connect to Google calendar : [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118 auth_credentials: Invalid username or password for CalDAV calendar 'cal1' [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal1, request REPORT to /calendar/dav/i...@mydomain.tld/events/: Could not authenticate to server: rejected Basic challenge [cal1] type = caldav url = https://www.google.com/calendar/dav/i...@mydomain.tld/events/ user = i...@mydomain.tld secret = mysecret refresh = 15 timeframe = 60 When I go to the URL https://www.google.com/calendar/dav/i...@mydomain.tld/events/ I can log in with the credentials to the calendar (and get a download window for the calendar file). So it seems not a problem of authentication to me. But what then could be the real issue here ? Thanks Kind regards Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue priority not respected
Hello I notice that priority of queue members is not being respected. Using mysql realtime. These are the queue members (in table queue_members) : Local/queuemem0@ExternalCallFromQueue Local/queuemem1@ExternalCallFromQueue Local/queuemem2@ExternalCallFromQueue Local/queuemem3@ExternalCallFromQueue Local/queuemem4@ExternalCallFromQueue Local/queuemem5@ExternalCallFromQueue Local/queuemem6@ExternalCallFromQueue Asterisk queue show : sipserver*CLI> queue show myqueueq myqueueq has 0 calls (max unlimited) in 'ringall' strategy (10s holdtime, 99s talktime), W:0, C:13, A:0, SL:0.0% within 0s Members: queuemem0 (Local/queuemem0@ExternalCallFromQueue) with penalty 1 (realtime) (Not in use) has taken 5 calls (last was 551 secs ago) queuemem1 (Local/queuemem1@ExternalCallFromQueue) with penalty 1 (realtime) (Not in use) has taken no calls yet queuemem2 (Local/queuemem2@ExternalCallFromQueue) with penalty 2 (realtime) (Not in use) has taken 4 calls (last was 1314 secs ago) queuemem3 (Local/queuemem3@ExternalCallFromQueue) with penalty 2 (realtime) (Not in use) has taken 3 calls (last was 1408 secs ago) queuemem4 (Local/queuemem4@ExternalCallFromQueue) with penalty 4 (realtime) (Not in use) has taken 1 calls (last was 1937 secs ago) queuemem5 (Local/queuemem5@ExternalCallFromQueue) with penalty 4 (realtime) (Not in use) has taken no calls yet queuemem6 (Local/queuemem6@ExternalCallFromQueue) with penalty 3 (realtime) (Not in use) has taken no calls yet No Callers In verbose log I can see that queuemem6 with penalty 3 is not being contacted : [Oct 5 10:07:17] VERBOSE[21097] app_dial.c: [Oct 5 10:07:17] -- Called SIP/queuemem1 [Oct 5 10:07:17] VERBOSE[21098] app_dial.c: [Oct 5 10:07:17] -- Called SIP/queuemem0 --> busy [Oct 5 10:07:17] VERBOSE[21100] app_dial.c: [Oct 5 10:07:17] -- Called SIP/queuemem2 [Oct 5 10:07:17] VERBOSE[21099] app_dial.c: [Oct 5 10:07:17] -- Called SIP/queuemem3 --> busy [Oct 5 10:07:17] VERBOSE[21101] app_dial.c: [Oct 5 10:07:17] -- Called SIP/queuemem5 [Oct 5 10:07:17] VERBOSE[21102] app_dial.c: [Oct 5 10:07:17] -- Called SIP/queuemem4 queuemem0 and queuemem1 (priority 1) are busy, so queuemem2 and queuemem3 (priority 2) are being called. So far so good. But we placed queuemem2 and queuemem3 (priority 2) also busy. So queuemem6 should be rung as priority 3. What do we see : queuemem4 and queuemem5 (priority 4) are being called ! This not correct. queuemem6 is never contacted. Why are priorities here not beining respected ? Kind regards Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Queues : linear strategy WITH priority
On 12-08-15 16:31, A J Stiles wrote: On Wednesday 12 Aug 2015, Jonas Kellens wrote: Hello I was wondering of it is possible to have Queue Agents with the same priority (penalty) but with a certain order ? So I have 20 Agents. Agent 1 till Agent 10 has penalty 1. Agent 11 till Agent 15 has penalty 2. (only contacted if 1 -> 10 are busy) Agent 16 till Agent 20 has penalty 3. (only contacted if 1 -> 10 and 11 -> 15 are busy) Within the range of Agent 1 till Agent 10, can I have a certain order in these Agents in which they are rung ?? Like Agent 1 -> Agent 5 -> Agent 2 & 3 & 4 -> Agent 6 -> Agent 7 -> Agent 8 & 9 & 10. What's wrong with giving agent 1 penalty 1; agent 5 penalty 2; agents 2, 3 and 4 penalty 3; agent 6 penalty 4, agent 7 penalty 5, and so forth? By giving a different penalty to Agents 1 to 10, there is no order. With penalty, the Agent keeps on being contacted untill it takes the call. Many forget that this is how penalties work ! So in stead of going from Agent 1 to Agent 5 to Agent 2,3,4 it is very possible that Agent 5 keeps on ringing when Agent 1 is 'busy calling', in stead of going further to Agents 2,3,5. In your case, Agent 5 will be called over and over again untill it takes the call. Not exactly what I'm looking for. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Queues : linear strategy WITH priority
Hello I was wondering of it is possible to have Queue Agents with the same priority (penalty) but with a certain order ? So I have 20 Agents. Agent 1 till Agent 10 has penalty 1. Agent 11 till Agent 15 has penalty 2. (only contacted if 1 -> 10 are busy) Agent 16 till Agent 20 has penalty 3. (only contacted if 1 -> 10 and 11 -> 15 are busy) Within the range of Agent 1 till Agent 10, can I have a certain order in these Agents in which they are rung ?? Like Agent 1 -> Agent 5 -> Agent 2 & 3 & 4 -> Agent 6 -> Agent 7 -> Agent 8 & 9 & 10. I guess I need 'linear' strategy, but will penalty option still work ? Thank you for your feedback. Kind regards. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes
On 07-08-15 13:23, Ethy H. Brito wrote: On Fri, 07 Aug 2015 12:47:40 +0200 Jonas Kellens wrote: Hello I have 2 strange errors when using the Background()-application and DTMF-input that is received. First of all, my first 2 lines are not being executed. The first line being executed is the Set() application, thus line 3. Secondly, the received digits (911) is not the same as the EXTEN (which is set to 91). exten => ivr,n,Set(TIMEOUT(digit)=2) exten => ivr,n,Background(/var/lib/asterisk/sounds/${ASTPROMPT}) exten => _X.,1,NoOp() exten => _X.,n,NoOp(input=${EXTEN}) exten => _X.,n,Set(choice=${EXTEN}) [Aug 7 12:31:26] -- Executing [ivr@pbx-routing:7] Set("SIP/SipAgenT-0626", "TIMEOUT(digit)=2") in new stack [Aug 7 12:31:26] -- Digit timeout set to 2.000 [Aug 7 12:31:26] -- Executing [ivr@pbx-routing:8] BackGround("SIP/SipAgenT-0626", "/var/lib/asterisk/sounds/5003") in new stack [Aug 7 12:31:26] -- Playing '/var/lib/asterisk/sounds/5003.slin' [Aug 7 12:31:41] NOTICE[3886]: ast_expr2.y:763 compose_func_args: argbuf allocated 4 bytes; [Aug 7 12:31:41] NOTICE[3886]: ast_expr2.y:782 compose_func_args: argbuf uses 3 bytes; [Aug 7 12:31:41] -- Executing [911@pbx-routing:1] Set("SIP/SipAgenT-0626", "choice=91") in new stack I have reloaded the dialplan several times, but the first 2 lines never get executed. In stead they generate the error : ast_expr2.y:763 compose_func_args: argbuf allocated 4 bytes; Anyone know what is going on here ? Kind regards, Jonas. Hi Jonas What is the output from "dialplan show" for this particular piece of code? cheers Ethy Hello "dialplan show" shows the following : '_X.' => 1. NoOp() [pbx_config] 2. NoOp(input=${EXTEN}) [pbx_config] 3. Set(choice=${EXTEN}) [pbx_config] But like I said, the first 2 lines do not get executed. I don't understand why if extension is 911 the code says : "choice=91" in this line : -- Executing [911@pbx-routing:1] Set("SIP/SipAgenT-0626", "choice=91") You see exten is 911, but when allocating ${EXTEN} to the variable "choice" it suddenly is 91 ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes
Hello I have 2 strange errors when using the Background()-application and DTMF-input that is received. First of all, my first 2 lines are not being executed. The first line being executed is the Set() application, thus line 3. Secondly, the received digits (911) is not the same as the EXTEN (which is set to 91). exten => ivr,n,Set(TIMEOUT(digit)=2) exten => ivr,n,Background(/var/lib/asterisk/sounds/${ASTPROMPT}) exten => _X.,1,NoOp() exten => _X.,n,NoOp(input=${EXTEN}) exten => _X.,n,Set(choice=${EXTEN}) [Aug 7 12:31:26] -- Executing [ivr@pbx-routing:7] Set("SIP/SipAgenT-0626", "TIMEOUT(digit)=2") in new stack [Aug 7 12:31:26] -- Digit timeout set to 2.000 [Aug 7 12:31:26] -- Executing [ivr@pbx-routing:8] BackGround("SIP/SipAgenT-0626", "/var/lib/asterisk/sounds/5003") in new stack [Aug 7 12:31:26] -- Playing '/var/lib/asterisk/sounds/5003.slin' [Aug 7 12:31:41] NOTICE[3886]: ast_expr2.y:763 compose_func_args: argbuf allocated 4 bytes; [Aug 7 12:31:41] NOTICE[3886]: ast_expr2.y:782 compose_func_args: argbuf uses 3 bytes; [Aug 7 12:31:41] -- Executing [911@pbx-routing:1] Set("SIP/SipAgenT-0626", "choice=91") in new stack I have reloaded the dialplan several times, but the first 2 lines never get executed. In stead they generate the error : ast_expr2.y:763 compose_func_args: argbuf allocated 4 bytes; Anyone know what is going on here ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with realtime mysql I can't seem to resolve
Hello I have already several Asterisk servers running with similar configuration, but now I stumble into a problem. I have mysql configuration res_config_mysql.conf : [MyAsteriskDB] dbhost = 127.0.0.1 dbname = MyAsteriskDB dbuser = astadmin dbpass = mysecret dbport = 3306 dbsock = /var/lib/mysql/mysql.sock requirements=warn ; or createclose or createchar Realtime seems to be loaded : *CLI> realtime mysql status general configured for asterisk on socket file /var/lib/mysql/mysql.sock with username asterisk. MyAsteriskDB connected to MyAsteriskDB@127.0.0.1, port 3306 with username astadmin for 12 minutes. [May 22 10:32:02] ERROR[11269]: res_config_mysql.c:1599 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on /var/lib/mysql/mysql.sock (err 1045). Check debug for more info. *CLI> However, SIP-registration for SIP peer can not be found : [May 22 10:32:50] NOTICE[11077]: chan_sip.c:24957 handle_request_register: Registration from '' failed for '11.22.33.44:5060' - No matching peer found Debug logs say : [May 22 10:48:06] DEBUG[11077] res_config_mysql.c: MySQL RealTime: Connection okay. [May 22 10:48:06] DEBUG[11077] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = 'testacc66' AND host = 'dynamic' [May 22 10:48:06] DEBUG[11077] res_config_mysql.c: MySQL RealTime: Connection okay. [May 22 10:48:06] DEBUG[11077] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = 'testacc66' But sip peer testacc66 really exists in my database in table sip_buddies. It can not be found ?! What else is there for me to investigate ? Can u help me ? Thanks ! Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use dialplan variables from MySQL database and replace with value
Hello i have the following field (text string) in a MySQL database : "${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4}" I read this string form the database and want to have the dialplan variables to be replaced with the correct content. How can I do this ? Currently this is not working. The variable ${PARAMS} contains the exact string of the database field : my dialplan : exten => s,n,MYSQL(Connect connid localhost dbuser dbpass MyTable) exten => s,n,MYSQL(Query resultid ${connid} SELECT script_url, script_params FROM my_tbl WHERE ID="${myID}") exten => s,n,MYSQL(Fetch fetchid ${resultid} scriptURL PARAMS) exten => s,n,NoOp(scriptURL = ${scriptURL} PARAMS = ${PARAMS}) becomes : -- Executing [s@sub-details:4] MYSQL("SIP/SipT01-0012", "Connect connid localhost dbuser dbpass MyTable") in new stack -- Executing [s@sub-details:5] MYSQL("SIP/SipT01-0012", "Query resultid 1 SELECT script_url, script_params FROM my_tbl WHERE ID="2"") in new stack -- Executing [s@sub-details:6] MYSQL("SIP/SipT01-0012", "Fetch fetchid 2 scriptURL PARAMS") in new stack -- Executing [s@sub-details:7] NoOp("SIP/SipT01-0012", "scriptURL = call_end.php PARAMS = ${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4}") in new stack If the variable ${PARAMS} contains other variables " ${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4} ", how can I use the values of these variables in my dialplan ?? I want to use "${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4} " as input to my script inside my dialplan : -- Executing [h@pbx-routing:43] System("SIP/SipT01-0012", "/usr/bin/php /var/lib/asterisk/agi-bin/call_end.php ${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4}") in new stack But in stead of having ${KNUMMER} I want to have "112233", and in stead of having ${phone_number_to} I want to have "31023456789" and so on... Is this possible ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] park()-command always parks on default 701
Hello, I have the following in my dialplan : exten => callpark,n,Set(PARKINGDYNPOS=200-210) exten => callpark,n,Set(PARKINGDYNCONTEXT=parked_001) exten => callpark,n,Park(2s,parkinglot_001) I see on the CLI : [Nov 25 15:08:47] -- Executing [callpark@pbx-routing:10] Set("SIP/SipT01-000b", "PARKINGDYNPOS=200-210") in new stack [Nov 25 15:08:47] -- Executing [callpark@pbx-routing:11] Set("SIP/SipT01-000b", "PARKINGDYNCONTEXT=parked_001") in new stack [Nov 25 15:08:47] -- Executing [callpark@pbx-routing:12] Park("SIP/SipT01-000b", "5s,parkinglot_001") in new stack [Nov 25 15:08:47] == Parked SIP/SipT01-000b on 701 (lot parkinglot_001). Will timeout back to extension [pbx-routing] s, 1 in 50 seconds [Nov 25 15:08:47] -- Added extension '701' priority 1 to parked_77 Why does Asterisk park on 701 ? Why not on 200 ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue log realtime mysql
On 04-11-14 11:52, Jonas Kellens wrote: On 04-11-14 11:50, Ishfaq Malik wrote: On 4 November 2014 10:40, Jonas Kellens <mailto:jonas.kell...@telenet.be>> wrote: Hello, I have 5 Asterisk servers all using mysql realtime to store queue log information. There is 1 out of 5 servers which stores the data in 4 columns : 'data1' --> 'data 5'. All other servers store data in 1 column 'data' with the data seperated by pipe. I see no difference in my configuration of extconfig.conf and logger.conf. Maybe a hidden default value ? Can someone tell me which setting makes the mysql realtime driver store data in 1 column or in seperate columns ? Using Asterisk 1.8.12.2 Kind regards, Jonas. Are you using mysql_realtime or odbc with a mysql back end? Using mysql_realtime, not using odbc. Hello, is there any more feedback on this ? I still haven't found the difference in realtime configuration between this 1 server and my 4 other servers. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue log realtime mysql
On 04-11-14 11:50, Ishfaq Malik wrote: On 4 November 2014 10:40, Jonas Kellens <mailto:jonas.kell...@telenet.be>> wrote: Hello, I have 5 Asterisk servers all using mysql realtime to store queue log information. There is 1 out of 5 servers which stores the data in 4 columns : 'data1' --> 'data 5'. All other servers store data in 1 column 'data' with the data seperated by pipe. I see no difference in my configuration of extconfig.conf and logger.conf. Maybe a hidden default value ? Can someone tell me which setting makes the mysql realtime driver store data in 1 column or in seperate columns ? Using Asterisk 1.8.12.2 Kind regards, Jonas. Are you using mysql_realtime or odbc with a mysql back end? Using mysql_realtime, not using odbc. Kind regards, Jonas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users