Re: [asterisk-users] Attempting to get BLF working with linphone
On 12/06/2020 16:19, Olivier wrote: It seems a new Linphone 4.2 is to be published next week ! Hopefully, ... 1. its call history is useless to me, it works very poorly with sip proxying (i.e. asterisk), the design is clunky (no simple list of all calls) 2. it has no simple busy light support, only the complicated presence stuff that asterisk doesn't really support well and needs manual intervention. 3. the Qt interface is "prettier" than the old gtk2 interface but way less complete for telephone functions. -- _ -- 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 hints can be in multiple states; most sip NOTIFY dialogs only send one state
On 10/06/2020 15:40, Joshua C. Colp wrote: You wouldn't be able to access such information from ast_sip_presence_exten_state_to_str, that function is strictly for taking in instructions/data and producing the output. The user of it would need to pass in a value to turn on this new behavior. From that level the ast_sip_exten_state_data structure can optionally have a subscription, which itself has the endpoint that was used to establish the subscription. Ok, I'll look at that when I get around to moving to chan_pjsip. I'm very slow at changing working configurations. :) -- _ -- 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 hints can be in multiple states; most sip NOTIFY dialogs only send one state
Asterisk can know that one of the attached phones is both "ringing" and "on the phone". However the sip NOTIFY it sends out to interested parties can only communicate one state, for example with pidf+xml it can either send "Ringing" or "On the phone" and so it sends "Ringing". This makes the "busy lights" less than useful, if a call makes multiple phones ring you can't tell, looking at the busy lights, which ones are busy, and so less likely to answer. In the chan_sip configuration there is an option "notifyringing": notifyringing *notifyringing* enables or disables notifications for the RINGING state when an extension is already INUSE. Only affects subscriptions using the *dialog-info* event package. Option can be configured in the general section only. It cannot be set per-peer. As the doc says this only applies to dialog-info style NOTIFY, not the pidf+xml format my phones use. Here is a patch that makes notifyringing work for pidf+xml. Generalising it for other formats is left as an exercise for the reader. Of course chan_sip is obsolete. How might this be done for chan_pjsip? Parts of the code are similar, but the layering is vastly different. How could the ast_sip_presence_exten_state_to_str function in res/res_pjsip/presence_xml.c get at the pjsip configuration? Description: make "notifyringing" work with pidf+xml If sip config specifies notifyringing=no and an extension is in a call then we send out "On a call" instead of "Ringing" so people can see who is not going to pick the call up. Author: John Hughes Last-Update: 2020-06-09 --- asterisk-13.14.1~dfsg.orig/channels/chan_sip.c +++ asterisk-13.14.1~dfsg/channels/chan_sip.c @@ -14966,7 +14966,10 @@ static void state_notify_build_xml(struc statestring = (sip_cfg.notifyringing) ? "early" : "confirmed"; local_state = NOTIFY_INUSE; pidfstate = "busy"; - pidfnote = "Ringing"; + if (subscribed == PIDF_XML && !sip_cfg.notifyringing) + pidfnote = "On the phone"; + else + pidfnote = "Ringing"; break; case AST_EXTENSION_RINGING: statestring = "early"; -- _ -- 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] Attempting to get BLF working with linphone
On 26/05/2020 15:33, Olivier wrote: Hi John, 1. Could you get any further, in your quest for working BLF with linphone ? The patches to get linphone-3.12 BLF working with Asterisk are here: http://perso.calvaedi.com/~john/linphone-3/ They're pretty damnned trivial: 1. add the "Accept" header to the SUBSCRIBE message so asterisk doesn't reject it. 2. don't trash the SIP dialog if the SUBSCRIBE refresh is rejected because of a stale nonce. 3. If asterisk says the user is on the phone set the status to on the phone. All except the 3rd one are compatible with linphone-4. Implementing the same feature with linphone-4 is left as an exercise for the reader. 2. Have you tried with a different Linphone version (4.12 is pending on Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ? Version 4 of linphone is, frankly, rubbish. I have managed to hack it to the point where presence shows green for connected contact and grey for disconnected. However this requires setting the "send subscribe" flag in the linphone contacts db and linphone 4 has no UI for setting this flag, you have to do it using sqlite3 directly (or setting up your contacts in linphone 3). -- _ -- 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] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
On 14/05/2020 16:41, Joshua C. Colp wrote: On Thu, May 14, 2020 at 11:31 AM John Hughes <mailto:j...@calva.com>> wrote: On 14/05/2020 08:10, John Hughes wrote: I am having a problem with one of my callers who is using either g729 or alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up. Why? Last minute thought: Is it possible that the caller is sending g729 in RTP even though the SIP negotiation clearly chooses alaw? Maybe I need some RTP debugging. And in fact that is exactly what's happening. And when I look at the RTP debugging I see the data from the remote is: Got RTP packet from xx.xx.xx.xx:50644 (type 18, seq 001338, ts 610458, len 20) AAArgh! Type 18 is g729. Why on earth is the remote sending me g729 when I clearly said the only thing I could do was alaw. Is this legal? Is the other side broken? It shouldn't be sending it, but as well we should be ignoring it. I believe we do ignore in modern versions, I can't speak for your old one. As for why... I don't really have an answer. Ok, so maybe upgrading my asterisk would be a good idea, but I don't think it'll fix this problem, they sent me 6 g729 packets before the communication was cut, I'm pretty sure they've just ignored the results of the negotiation. I hope I can get them to fix their system... -- _ -- 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] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
On 14/05/2020 08:10, John Hughes wrote: I am having a problem with one of my callers who is using either g729 or alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up. Why? Last minute thought: Is it possible that the caller is sending g729 in RTP even though the SIP negotiation clearly chooses alaw? Maybe I need some RTP debugging. And in fact that is exactly what's happening. <--- SIP read from UDP:SUPPLIER:5060 ---> INVITEsip:LOCAL@ASTERISK:5060 SIP/2.0 Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9 From:;tag=gK02498cb1 To: Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: P-Asserted-Identity: Supported: timer,100rel,precondition Session-Expires: 1800 Min-SE: 90 Content-Length: 282 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 176880 320591 IN IP4 SUPPLIER s=SIP Media Capabilities c=IN IP4 213.41.124.6 t=0 0 m=audio 8526 RTP/AVP 18 8 101 *a=rtpmap:18 G729/8000* a=fmtp:18 annexb=no *a=rtpmap:8 PCMA/8000* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 <-> So he says he wants g729 or alaw Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw|gsm), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (*alaw*) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) And asterisk calculates that the common codecs are just alaw, So asterisk says: "let's do alaw": <--- Reliably Transmitting (no NAT) to SUPPLIER:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER From:;tag=gK02498cb1 To:;tag=as4502927f Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 227409966 227409966 IN IP4 ASTERISK s=Asterisk PBX 13.14.1~dfsg-2+deb9u4 c=IN IP4 ASTERISK t=0 0 m=audio 13948 RTP/AVP 8 101 *a=rtpmap:8 PCMA/8000* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <> And when I look at the RTP debugging I see the data from the remote is: Got RTP packet from xx.xx.xx.xx:50644 (type 18, seq 001338, ts 610458, len 20) AAArgh! Type 18 is g729. Why on earth is the remote sending me g729 when I clearly said the only thing I could do was alaw. Is this legal? Is the other side broken? -- _ -- 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] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
On 14/05/2020 16:08, Michael L. Young wrote: I am having a problem with one of my callers who is using either g729 or alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up. Why? Last minute thought: Is it possible that the caller is sending g729 in RTP even though the SIP negotiation clearly chooses alaw? Maybe I need some RTP debugging. Asterisk 13.14.1 on Debian, using chan_sip. Hi John, Maybe a newer version of Asterisk would help? The latest release for 13 is version 13.33. The version you are on was released 3 years ago. Well, like I said I'm on Debian, using the packaged version. If I want to upgrade I'll have to compile it myself, or upgrade to Debian buster to get 16.2.1 Here is an issue which looks like what you describe and was fixed in 13.16 https://issues.asterisk.org/jira/browse/ASTERISK-26143 That doesn't seem to be the same problem. My problem is that the other end is sending g729, which my asterlsk can't do at all, and tells the remote it can't do. I'm shocked that the remote is trying to send me stuff using a codec that I didn't say I could handle. -- _ -- 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] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
I am having a problem with one of my callers who is using either g729 or alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up. Why? Last minute thought: Is it possible that the caller is sending g729 in RTP even though the SIP negotiation clearly chooses alaw? Maybe I need some RTP debugging. Asterisk 13.14.1 on Debian, using chan_sip. Here's the trace: <--- SIP read from UDP:SUPPLIER:5060 ---> INVITEsip:LOCAL@ASTERISK:5060 SIP/2.0 Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9 From:;tag=gK02498cb1 To: Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: P-Asserted-Identity: Supported: timer,100rel,precondition Session-Expires: 1800 Min-SE: 90 Content-Length: 282 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 176880 320591 IN IP4 SUPPLIER s=SIP Media Capabilities c=IN IP4 213.41.124.6 t=0 0 m=audio 8526 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 <-> --- (17 headers 13 lines) --- Sending to SUPPLIER:5060 (no NAT) Sending to SUPPLIER:5060 (no NAT) Using INVITE request as basis request - 205665777_90679951@SUPPLIER Found peer 'supplier' for 'REMOTE' from SUPPLIER:5060 Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw|gsm), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 213.41.124.6:8526 Looking for LOCAL in supplier-in (domain ASTERISK) sip_route_dump: route/path hop: So, all looking good here, we've worked out that the combined capabilities are (alaw) <--- Transmitting (no NAT) to SUPPLIER:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER From:;tag=gK02498cb1 To: Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <> <--- Transmitting (no NAT) to SUPPLIER:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER From:;tag=gK02498cb1 To:;tag=as4502927f Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <> Audio is at 13948 Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to SUPPLIER:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER From:;tag=gK02498cb1 To:;tag=as4502927f Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 227409966 227409966 IN IP4 ASTERISK s=Asterisk PBX 13.14.1~dfsg-2+deb9u4 c=IN IP4 ASTERISK t=0 0 m=audio 13948 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <> And that's good to, we've sent the OK for the INVITE saying that we want alaw. <--- SIP read from UDP:SUPPLIER:5060 ---> ACKsip:LOCAL@ASTERISK:5060 SIP/2.0 Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5bc037285f864da9 From:;tag=gK02498cb1 To:;tag=as4502927f Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 ACK Max-Forwards: 70 Content-Length: 0 <-> --- (8 headers 0 lines) --- [May 13 13:46:58] WARNING[7245][C-31da]: channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw) What's this nonsense! Why is set_format trying to use g729! Scheduling destruction of SIP dialog '205665777_90679951@SUPPLIER' in 32000 ms (Method: ACK) set_destination: Parsing for address/port
Re: [asterisk-users] Attempting to get BLF working with linphone
On 23/03/2020 18:51, Joshua C. Colp wrote: On Mon, Mar 23, 2020 at 2:45 PM John Hughes <mailto:j...@calva.com>> wrote: Why is asterisk giving an error 500? I can find no reason, there is nothing in any log. The sequence number is from the past. The first SUBSCRIBE is sequence number 22 (check the CSeq header). The second is 20. The third is 21. It appears as though this is from the past, so it receives a 500. Ok, I've had some back and forth with the linphone developers and they contend that although the sequence number on the 2nd and 3rd SUBSCRIBE messages start a new sequence this is legal as it is a new conversation -- the "tag=" on the From has changed. Are they right? (Notice that the tag= from asterisk also changes). <--- SIP read from UDP:10.27.128.3:5060 <http://10.27.128.3:5060> ---> SUBSCRIBE sip:jacques@10.27.128.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;rport From: ;*tag=iGH81k5xf* To: ;tag=as3c7de68c CSeq: 22 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) Authorization: Digest realm="asterisk", nonce="188b095b", algorithm=MD5, username="john", uri="sip:jacques@10.27.128.1:5060", response="bdbc7cbac4453fd643050bf28996a68e" <-> --- (14 headers 0 lines) --- Found peer 'john' for 'john' from 10.27.128.3:5060 <http://10.27.128.3:5060> <--- Transmitting (no NAT) to 10.27.128.3:5060 <http://10.27.128.3:5060> ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;received=10.27.128.3;rport=5060 From: ;*tag=iGH81k5xf* To: ;tag=as3c7de68c Call-ID: SQOclJgm4O CSeq: 22 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3144c0a9", stale=true Content-Length: 0 <> Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.27.128.3:5060 <http://10.27.128.3:5060> ---> SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;rport From: ;*tag=c3Wvuu2XH <= new conversation* To: sip:jacq...@masked.masked.com CSeq: *20 SUBSCRIBE <=== sequence restarts* Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) <-> --- (13 headers 0 lines) --- Sending to 10.27.128.3:5060 <http://10.27.128.3:5060> (no NAT) Creating new subscription Sending to 10.27.128.3:5060 <http://10.27.128.3:5060> (no NAT) sip_route_dump: route/path hop: Found peer 'john' for 'john' from 10.27.128.3:5060 <http://10.27.128.3:5060> <--- Transmitting (no NAT) to 10.27.128.3:5060 <http://10.27.128.3:5060> ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;received=10.27.128.3;rport=5060 From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com;tag=as007ffc64 Call-ID: SQOclJgm4O CSeq: 20 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4224acfb" Content-Length: 0 <> Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.27.128.3:5060 <http://10.27.128.3:5060> ---> SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;rport From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com CSeq: 21 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) Authorization: Digest realm="asterisk", nonce="4224acfb", algorithm=MD5, username="john", uri="sip:jacq...@masked.masked.com", response="eb30a9801e78d2cb2c58c61200c50cb1" <-> --- (14 headers 0 lines) --- <--- Transmitting (no NAT) to 10.27.128.3:5060 <http://10.27.128.3:5060> ---> *SIP/2.0 500 Server error* Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;received=10.27.128.3;rport=5060 From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com;tag=as3c7de68c Call-ID: SQOclJgm4O CSeq: 21 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow:
Re: [asterisk-users] Attempting to get BLF working with linphone
On 23/03/2020 18:51, Joshua C. Colp wrote: On Mon, Mar 23, 2020 at 2:45 PM John Hughes <mailto:j...@calva.com>> wrote: Why is asterisk giving an error 500? I can find no reason, there is nothing in any log. The sequence number is from the past. The first SUBSCRIBE is sequence number 22 (check the CSeq header). The second is 20. The third is 21. It appears as though this is from the past, so it receives a 500. Why does asterisk think the error 500 is going to be acked? It doesn't. The message is for something else, it refers to sequence number 103. Ok, thanks, that's clear and obvious. Now I have to go beat up(*) the linphone people. ((*) in the nicest possible way of course). -- _ -- 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] Attempting to get BLF working with linphone
So I've got a bit further with my project to get BLF working between asterisk and linphone. Initially asterisk was rejecting linphone's SUBSCRIBE messages because they didn't have an Accept: header. I've fixed that and now the initial SUBSCRIBE messages work and I see all my online contacts in green. But after a few minutes linphone attempts to renew the subscriptions and asterisk is not happy at all: <--- SIP read from UDP:10.27.128.3:5060 ---> SUBSCRIBE sip:jacques@10.27.128.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;rport From: ;tag=iGH81k5xf To: ;tag=as3c7de68c CSeq: 22 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) Authorization: Digest realm="asterisk", nonce="188b095b", algorithm=MD5, username="john", uri="sip:jacques@10.27.128.1:5060", response="bdbc7cbac4453fd643050bf28996a68e" <-> --- (14 headers 0 lines) --- Found peer 'john' for 'john' from 10.27.128.3:5060 <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;received=10.27.128.3;rport=5060 From: ;tag=iGH81k5xf To: ;tag=as3c7de68c Call-ID: SQOclJgm4O CSeq: 22 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3144c0a9", stale=true Content-Length: 0 <> Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.27.128.3:5060 ---> SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;rport From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com CSeq: 20 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) <-> --- (13 headers 0 lines) --- Sending to 10.27.128.3:5060 (no NAT) Creating new subscription Sending to 10.27.128.3:5060 (no NAT) sip_route_dump: route/path hop: Found peer 'john' for 'john' from 10.27.128.3:5060 <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;received=10.27.128.3;rport=5060 From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com;tag=as007ffc64 Call-ID: SQOclJgm4O CSeq: 20 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4224acfb" Content-Length: 0 <> Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.27.128.3:5060 ---> SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;rport From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com CSeq: 21 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) Authorization: Digest realm="asterisk", nonce="4224acfb", algorithm=MD5, username="john", uri="sip:jacq...@masked.masked.com", response="eb30a9801e78d2cb2c58c61200c50cb1" <-> --- (14 headers 0 lines) --- <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> *SIP/2.0 500 Server error* Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;received=10.27.128.3;rport=5060 From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com;tag=as3c7de68c Call-ID: SQOclJgm4O CSeq: 21 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <> [Mar 23 18:23:09] WARNING[2128]: chan_sip.c:4071 retrans_pkt: Retransmission timeout reached on transmission SQOclJgm4O for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response Why is asterisk giving an error 500? I can find no reason, there is nothing in any log. Why does asterisk think the error 500 is going to be acked? -- _ -- 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:
Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH
On 23/03/2020 11:29, Joshua C. Colp wrote: On Mon, Mar 23, 2020 at 7:15 AM John Hughes <mailto:j...@calva.com>> wrote: Hi, in these dark days of COVID-19 lockdown I'm using linphone to connect to my office asterisk system for working from home. It's going pretty well but the presence/BLF functions don't appear to work. In the linphone logs and asterisk debug I find that asterisk is rejecting linphone's PUBLISH message: Asterisk has no support for receiving/storing/using such a PUBLISH message. Asterisk instead generates state itself based on whether something is on the phone, busy, etc. This is received using a SUBSCRIBE and NOTIFY. Aha! Thanks a bunch. Now I just have to fix linphone's broken SUBSCRIBE... Mar 23 11:48:37] WARNING[2128]: chan_sip.c:28198 handle_request_subscribe: SUBSCRIBE failure:*no Accept header*: pvt: stateid: -1, laststate: 0, dialogver: 0, subscribecont: '', subscribeuri: '' -- _ -- 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] SIP/2.0 489 Bad Event in reply to a PUBLISH
Hi, in these dark days of COVID-19 lockdown I'm using linphone to connect to my office asterisk system for working from home. It's going pretty well but the presence/BLF functions don't appear to work. In the linphone logs and asterisk debug I find that asterisk is rejecting linphone's PUBLISH message: <--- SIP read from UDP:10.27.128.3:5060 ---> PUBLISH sip:j...@xxx.xxx.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;rport From: ;tag=ZtFgBTxUL To: sip:j...@xxx.xxx.com CSeq: 20 PUBLISH Call-ID: SMHLUSLJD6 Max-Forwards: 70 Supported: replaces, outbound Event: presence Accept: application/pidf+xml Content-Length: 511 Content-Type: application/pidf+xml Expires: 3600 User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:pidfonline="http://www.linphone.org/xsds/pidfonline.xsd; entity="sip:j...@xxx.xxx.com" xmlns="urn:ietf:params:xml:ns:pidf"> open sip:j...@xxx.xxx.com 2020-03-23T09:40:43Z <-> --- (14 headers 3 lines) --- Sending to 10.27.128.3:5060 (no NAT) <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;received=10.27.128.3;rport=5060 From: ;tag=ZtFgBTxUL To: sip:j...@xxx.xxx.com;tag=as674d428f Call-ID: SMHLUSLJD6 CSeq: 20 PUBLISH Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 I can find nothing in the asterisk logs that says *why* it doesn't like the publish. Help? -- _ -- 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] Distributed FAX - How to best complement asterisk ?
Benny Amorsen wrote: PvK == Philipp von Klitzing [EMAIL PROTECTED] writes: PvK Some of the bigger MFC printer/copy/fax combo devices by Brother PvK (and maybe also other vendors?) provide a fax-via-smtp feature PvK and can built fax networks that way. As far as I can tell, the Brother boxes require the user to enter fax-over-internet addresses as actual email addresses. There is no way to tell the machine to always use fax-over-internet, and there is also no way to configure a default domain. Asking fax users to select email and then type in [EMAIL PROTECTED] on a tiny little keyboard just won't fly. I don't understand why no fax vendors offer an easy fax-via-email user interface. And why on earth don't all the HP MFC's that have Ethernet and or Wireless and a Fax modem also have T.38 built in? I've got 2 of the damn things and the fax capability is useless to me. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN2#02: too much voice to send for NCCI=0x40502
My problem: Sometimes the sound seems to cut on calls in progress. We (on our local SIP phones, Thomson ST2030's) can't hear the remote caller. The caller may hear some kind of horrid sklurk and then it goes dead for them too. Our Asterisk is connected to the France Telecom network by an Eicon Diva Server 4xBRI (we have 3 BRI lines). Here's what I've found by turning on huge amounts of debug stuff: everything runs just tickety-boo until the remote end seems to stop sending us packets, our packets seem not to be sent and we get the infamous too much voice to send error. Help! What might be going on? Here's an example: [ All is well: ] Got RTP packet from192.168.6.185:41000 (type 08, seq 018498, ts 1479920, len 80) Got RTP packet from192.168.6.185:41000 (type 08, seq 018499, ts 148, len 80) DATA_B3_REQ ID=002 #0xee37 LEN=0030 Controller/PLCI/NCCI= 0x40502 Data32 = 0x0 DataLength = 0xa0 DataHandle = 0xb624 Flags = 0x0 Data64 = 0x82fa70 DATA_B3_CONF ID=002 #0xee37 LEN=0016 Controller/PLCI/NCCI= 0x40502 DataHandle = 0xb624 Info= 0x0 DATA_B3_IND ID=002 #0x8a51 LEN=0030 Controller/PLCI/NCCI= 0x40502 Data32 = 0x0 DataLength = 0xa0 DataHandle = 0x49 Flags = 0x0 Data64 = 0x2aaac7caa49e DATA_B3_RESP ID=002 #0x8a51 LEN=0014 Controller/PLCI/NCCI= 0x40502 DataHandle = 0x49 -- ISDN2#02: DATA_B3_IND (len=160) fr.datalen=160 fr.subclass=8 Sent RTP packet to 192.168.6.185:41000 (type 08, seq 064338, ts 1428080, len 000160) [ but at this point we seem to stop getting data from the network. We continue to send: ] Got RTP packet from192.168.6.185:41000 (type 08, seq 018500, ts 1480080, len 80) Got RTP packet from192.168.6.185:41000 (type 08, seq 018501, ts 1480160, len 80) DATA_B3_REQ ID=002 #0xee39 LEN=0030 Controller/PLCI/NCCI= 0x40502 Data32 = 0x0 DataLength = 0xa0 DataHandle = 0xb625 Flags = 0x0 Data64 = 0x82fb50 DATA_B3_CONF ID=002 #0xee39 LEN=0016 Controller/PLCI/NCCI= 0x40502 DataHandle = 0xb625 Info= 0x0 [... 8 RTP packets, 4 DATA_B3_REQ's omitted ] [ And then: ] Got RTP packet from192.168.6.185:41000 (type 08, seq 018510, ts 1480880, len 80) Got RTP packet from192.168.6.185:41000 (type 08, seq 018511, ts 1480960, len 80) ISDN2#02: too much voice to send for NCCI=0x40502 Got RTP packet from192.168.6.185:41000 (type 08, seq 018512, ts 1481040, len 80) Got RTP packet from192.168.6.185:41000 (type 08, seq 018513, ts 1481120, len 80) ISDN2#02: too much voice to send for NCCI=0x40502 [... Someone gets fed up and cuts the B channel: ] Got RTP packet from192.168.6.185:41000 (type 08, seq 018610, ts 140, len 80) Got RTP packet from192.168.6.185:41000 (type 08, seq 018611, ts 1488960, len 80) ISDN2#02: too much voice to send for NCCI=0x40502 DISCONNECT_B3_IND ID=002 #0x8a8a LEN=0015 Controller/PLCI/NCCI= 0x40502 Reason_B3 = 0x0 NCPI= default DISCONNECT_B3_RESP ID=002 #0x8a8a LEN=0012 Controller/PLCI/NCCI= 0x40502 End of conversation. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Horrible problem - calls losing sound
We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI). We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian system, with chan-capi 1.0.1. Any idea what could be going wrong, where to look c? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
Tzafrir Cohen wrote: On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote: Just as you have rtp debug, you have bri debug . bri debug span 1 and hope for a friendly ISDN guru on the list... Does nothing for me - my isdn is connected via chan-capi. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
Tzafrir Cohen wrote: On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote: Hi John! I have the same problem, the system contains 1 port Billion ISDN BRI card, and 1 sip trunk. This is a trixbox with Asterisk 1.2.22-BRIstuffed-0.3.0-PRE-1y-i The ISDN call is forwarded to a ring-group. The 6 sip phones are welltech lp399 series. If incoming the call get wrong, we can not hear the other side, but they hear us. In my case the rtp debug shows there are no incoming rtp packets from asterisk to SIP phone. If somebody experienced this problem, please help US! Just as you have rtp debug, you have bri debug . bri debug span 1 and hope for a friendly ISDN guru on the list... Ah, I can get the same (excessive!) info by set verbose 8 capi debug ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
Patrick wrote: On Thu, 2007-09-20 at 12:49 +0200, John Hughes wrote: We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI). We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian system, with chan-capi 1.0.1. Any idea what could be going wrong, where to look c? I'm don't know what is causing your problem but afaik you don't need bristuff with your Eicon Diva Server card. I have several asterisk 1.2.24 boxes with 1x BRI and 4x BRI Eicon Diva Server cards and I only use asterisk + chan_capi. This setup is very stable and works great. Perhaps you could try rebuilding asterisk without the bristuff patch then rebuild and install chan_capi and see if the problem goes away? Yeah, I was being lazy, just using the asterisk from the Debian repositories. Guess I'll try with my own build. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt
Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt
John Hughes wrote: Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11 Ah, I'm so dumb - it's this bit: * Queues depend on the channel driver reporting the proper state for each member of the queue. To get proper signalling on queue members that use the SIP channel driver, you need to enable a call limit (could be set to a high value so it is not put into action) and also make sure that both inbound and outbound calls are accounted for. Example: [general] limitonpeer = yes [peername] type=friend call-limit=10 Could have sworn I had the call-limit, but in fact it was missing for the extension I was testing from! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SMS in France - allways get NAK
I'm trying to send an sms: smsq --motx-channel=CAPI/g1/0809101000 060739 X It seems to try to do something, but FT aren't happy: -- Attempting call on CAPI/g1/0809101000 for application SMS(0) (Retry 1) == ISDN4#02: Setting up DTMF detector (PLCI=0x104, flag=1) [Sep 13 15:45:50] WARNING[23584]: pbx.c:5142 ast_pbx_outgoing_app2: CAPI/ISDN4#02/0809101000-1 already has a call detail record?? Channel CAPI/ISDN4#02/0809101000-1 was answered. Launching SMS(0) on CAPI/ISDN4#02/0809101000-1 -- SMS RX 93 00 6D -- SMS TX 91 0D 01 01 0A 81 60 70 93 66 66 00 F1 01 58 5C -- SMS RX 96 0A 01 A0 00 70 90 31 51 54 45 80 24 -- SMS TX 94 00 6C == ISDN4#02: CAPI Hangingup for PLCI=0x104 in state 2 [Sep 13 15:45:58] NOTICE[23584]: pbx_spool.c:371 attempt_thread: Call completed to CAPI/g1/0809101000 ISDN4#02: CAPI INFO 0x3495: Call rejected As I understand it the RX 96 is FT NAK'ing my TX 91 DELIVER. Any idea what I'm doing wrong? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS in France - allways get NAK
John Hughes wrote: I'm trying to send an sms: smsq --motx-channel=CAPI/g1/0809101000 060739 X It seems to try to do something, but FT aren't happy: -- Attempting call on CAPI/g1/0809101000 for application SMS(0) (Retry 1) == ISDN4#02: Setting up DTMF detector (PLCI=0x104, flag=1) [Sep 13 15:45:50] WARNING[23584]: pbx.c:5142 ast_pbx_outgoing_app2: CAPI/ISDN4#02/0809101000-1 already has a call detail record?? Channel CAPI/ISDN4#02/0809101000-1 was answered. Launching SMS(0) on CAPI/ISDN4#02/0809101000-1 -- SMS RX 93 00 6D -- SMS TX 91 0D 01 01 0A 81 60 70 93 66 66 00 F1 01 58 5C -- SMS RX 96 0A 01 A0 00 70 90 31 51 54 45 80 24 -- SMS TX 94 00 6C == ISDN4#02: CAPI Hangingup for PLCI=0x104 in state 2 [Sep 13 15:45:58] NOTICE[23584]: pbx_spool.c:371 attempt_thread: Call completed to CAPI/g1/0809101000 ISDN4#02: CAPI INFO 0x3495: Call rejected As I understand it the RX 96 is FT NAK'ing my TX 91 DELIVER. Any idea what I'm doing wrong? FT only accept the SMS if you use a valid sender address, adding the --motx-callerid= option fixed it. -- SMS RX 93 00 6D -- SMS TX 91 38 01 04 0A 81 60 58 70 11 11 00 F1 32 D3 B2 9B 0C AA CF 41 61 36 1B 94 7F D7 E5 A0 F6... -- SMS RX 95 09 01 00 70 90 31 51 95 25 80 A5 -- SMS TX 94 00 6C ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sort of OT: PBX vs CO
Jay R. Ashworth wrote: On Mon, Aug 13, 2007 at 02:57:09AM -0400, Alex Balashov wrote: On Sat, 11 Aug 2007, Jay R. Ashworth wrote: On Fri, Aug 10, 2007 at 05:20:38PM -0400, Alex Balashov wrote: It might be possible to glue something together with it and OpenSER and a media gateway control protocol like H.248 and a few of these SS7-IP appliances, but it would have all the ragtag qualities of Napoleon's army routed in Russia at the beginning of the 19th century. A class-5 switch is something I always wanted to program... You're certainly not the only one who has some enthusiasm for that. But using all this soft, high-level stuff to do it would be quite a chore, and the result very half-assed. At least, at this point in its evolution. Oh, I was actually thinking of Forth. :-) When I programmed switches it was in PO CORAL. These days I'd expect you to be doing it in ERLANG. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
Matthew J. Roth wrote: Recently, we were pushing our server to almost full CPU utilization. Since we've observed that Asterisk is CPU bound, we upgraded our server from a PowerEdge 6850 with four single-core Intel Xeon CPUs running at 3.16GHz, to a PowerEdge 6850 with 4 dual-core Intel Xeon CPUs running at 3.00GHz. The software installed is identical and a kernel build benchmark yielded promising results. The new dual-core server ran roughly 80% faster, which is about what we expected. As far as Asterisk is concerned, at low call volumes the dual-core server outperforms the single-core server at a similar rate. Outperforms in what sense? I'm working on a follow-up post that will demonstrate this with some benchmarks for a small number of calls in various scenarios on each machine. However, to our surprise as the number of concurrent calls increases, the performance gains begin to flatten out. In fact, it seems that somewhere between 200 and 300 calls, the two servers start to exhibit similar idle times despite one of them having twice as many cores. What do you mean by idle here? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
Sean M. Pappalardo wrote: Hi there. Just curious if you've checked out Linux clustering software such as OpenSSI ( http://www.openssi.org/ ) and run Asterisk on it? It features a multi-threaded cluster-aware shell (and custom kernel) that will automatically cluster-ize any regular Linux executable (such as the main Asterisk process.) If it works as advertised, it should just be a matter of adding boxes to the cluster to speed up processing. OpenSSI can't (at the moment) migrate threads between compute nodes. It can migrate separate processes, but doesn't Asterisk use threads? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks
Matthew J. Roth wrote: This post contains the benchmarks for Asterisk at low call volumes on similar single and dual-core servers. I'd appreciate it greatly if you took the time to read and comment on it. For me all these numbers look too small to be useful for benchmarking. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk (1.4) and hints/presence/BLF
Playing with hints/presence/BLF on asterisk I've made the following discoveries. 1. The wiki at http://www.voip-info.org/wiki/view/Asterisk+presence says: If you add incominglimit=1 to your peer in sip.conf, the SIP channel will notify you when that extension is busy. As incominglimit is obsolete you can use call-limit. Also you don't need to limit it to one, just having a call-limit at all works. (Tried with call-limit 20). What is the logic behind the linking of presence to call-limit? 2. A phone is only seen as busy if it's received an incoming call. Outgoing calls don't change the state. Why? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
