[asterisk-users] call failed... but why? What means SIP_ALREADYGONE?
Hi, I have watched a phenomen, that I can not explain... maybe one of you can see the reason why the call failed, and if the cause is the Snom Hardphone, or the asterisk, or the SIP-Provider... the debug log given below is all I have... What does Setting SIP_ALREADYGONE on dialog.. mean? thanks for watching, yves SIP Phone 110 (callerid 061444018110) tried to call the external Phone Number 0616677823 and gets an hangup after 2 seconds. Another try immediately after the failed call goes fine. The failed call did not arrive at the destination. [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Begin: parsing SIP Supported: timer, 100rel, replaces, from-change [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -timer- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: timer [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -100rel- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: 100rel [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -replaces- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: replaces [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -from-change- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: from-change [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.0.165:3072 [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Connection okay. [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '00616677823' AND h ost = 'dynamic' [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Connection okay. [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '00616677823' [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Stopping retransmission on '9a6bdc548d19-goay25ioz0nd' of Response 1: Match Found [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f2a74158788' [Feb 12 10:00:11] DEBUG[1567][C-380e] res_rtp_asterisk.c: Allocated port 19528 for RTP instance '0x7f2a74158788' [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: RTP instance '0x7f2a74158788' is setup and ready to go [Feb 12 10:00:11] DEBUG[1567][C-380e] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f2a74158788' [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Setting NAT on RTP to On [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP o=root 871055034 871055034 IN IP4 192.168.0.165... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.165... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 9 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 0 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 8 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 99 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 108 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 18 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 101 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:99 G726-32/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:108 AAL2-G726-32/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e]
[asterisk-users] Call failed becaus of SIP tanslate
Hi Guys, I have a the following issue when I ma trying to place a call through my voip provider, I am using an asterisk 1.2.21.1, do you have an idea what could fix this issue (as you can see when the other party answered, the call get dropped because of probably sip incompatibility) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) -- SIP/to-my-voip-11b955c0 answered SIP/8021-52514588 Nov 12 14:31:30 WARNING[21432]: channel.c:2781 ast_channel_make_compatible: No path to translate from SIP/8021-52514588(4) to SIP/to-my-voip-11b955c0(256) Nov 12 14:31:30 WARNING[21432]: app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/8021-52514588 compatible with SIP/to-my-voip-11b955c0 Thank you for any help! -- Abdullah -- _ -- 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 failed becaus of SIP tanslate
Ok, it seems like I don't have g729 codec intsalled, can I install this codec within asterisk 1.2 or just 1.4 and 1.6 are supported? On Fri, Nov 12, 2010 at 2:56 PM, khalid touati khalidtou...@gmail.comwrote: Hi Guys, I have a the following issue when I ma trying to place a call through my voip provider, I am using an asterisk 1.2.21.1, do you have an idea what could fix this issue (as you can see when the other party answered, the call get dropped because of probably sip incompatibility) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) -- SIP/to-my-voip-11b955c0 answered SIP/8021-52514588 Nov 12 14:31:30 WARNING[21432]: channel.c:2781 ast_channel_make_compatible: No path to translate from SIP/8021-52514588(4) to SIP/to-my-voip-11b955c0(256) Nov 12 14:31:30 WARNING[21432]: app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/8021-52514588 compatible with SIP/to-my-voip-11b955c0 Thank you for any help! -- Abdullah -- Abdullah -- _ -- 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 Failed Audio
Hi, On freepbx (GUI), whatever reason number fails we always get 'all circuits are busy' audio. Does anybody know how to get far end audio when we dial wrong number or when it's busy or unallocated number or failed with some other reason. Thanks, Deepika -- _ -- 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 Failed Audio
Sorry this is a list for the Asterisk GUI Project. I think you may have better luck on the FreePBX list / forums. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Mon, Oct 11, 2010 at 9:33 AM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi, On freepbx (GUI), whatever reason number fails we always get 'all circuits are busy' audio. Does anybody know how to get far end audio when we dial wrong number or when it’s busy or unallocated number or failed with some other reason. Thanks, Deepika -- _ -- 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] Call failed: 408 timeout
Hello: Here is my sip and extentions configuration and the log of x-lite, because i don`t can call inside my LAN with asterisk PBX 1.2 and i don`t have NAT. i hope you can help me. SIP.conf [default] include=anexos include=anexos1 include=anexos2 [anexos] exten= 100,1,Dial(SIP/100,0) exten= 100,2,Hangup [anexos1] exten= 101,1,Dial(SIP/101,0) exten= 101,2,Hangup [anexos2] exten= 102,1,Dial(SIP/102,0) exten= 102,2,Hangup EXTENTIONS.CONF bindport=5060 ; bindaddr=0.0.0.0 srvlookup=yes (10:23:26) : [100] type=friend secret= callerid=javier100 host=dynamic disallow=all allow=all context=default nat=no [101] type=friend secret= callerid=informatica101 host=dynamic disallow=all allow=all context=default nat=no [102] type=friend secret= callerid=admin102 host=dynamic disallow=all allow=all context=default nat=no LOG X-LITE (10:16:24) : © 2004 Xten Networks, Inc. All rights reserved. X-Lite release 1105d build stamp 9 License key: 31AC0B511918201B7ED760CE6BC073B6 Established SIP protocol listen on: 10.44.1.20:5060 Firewall Discovery Skipped SIP: 10.44.1.20:5060 RTP: 10.44.1.20:8000 NAT: 10.44.1.20 SEND TIME: 3079422292 SEND 0.0.0.100:5060 INVITE sip:100 SIP/2.0 Via: SIP/2.0/UDP 10.44.1.20:5060;rport;branch=z9hG4bK42C9872FBBB23E82A267FF9CA39FD454 From: informatica sip:1...@10.44.1.20;tag=93961341 To: sip:100 Contact: sip:1...@10.44.1.20:5060 Call-ID: 716c23be-d94f-060d-2f43-d9557f056...@10.44.1.20 CSeq: 41181 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 304 v=0 o=102 3079422269 3079422292 IN IP4 10.44.1.20 s=X-Lite c=IN IP4 10.44.1.20 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Attempting SIP protocol listen on: 10.44.1.20:5060 Established SIP protocol listen on: 10.44.1.20:5060 SEND TIME: 3079423989 SEND 0.0.0.100:5060 INVITE sip:100 SIP/2.0 Via: SIP/2.0/UDP 10.44.1.20:5060;rport;branch=z9hG4bK42C9872FBBB23E82A267FF9CA39FD454 From: informatica sip:1...@10.44.1.20;tag=93961341 To: sip:100 Contact: sip:1...@10.44.1.20:5060 Call-ID: 716c23be-d94f-060d-2f43-d9557f056...@10.44.1.20 CSeq: 41181 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 304 v=0 o=102 3079422269 3079422292 IN IP4 10.44.1.20 s=X-Lite c=IN IP4 10.44.1.20 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Attempting SIP protocol listen on: 10.44.1.20:5060 Established SIP protocol listen on: 10.44.1.20:5060 SEND TIME: 3079427009 SEND 0.0.0.100:5060 INVITE sip:100 SIP/2.0 Via: SIP/2.0/UDP 10.44.1.20:5060;rport;branch=z9hG4bK42C9872FBBB23E82A267FF9CA39FD454 From: informatica sip:1...@10.44.1.20;tag=93961341 To: sip:100 Contact: sip:1...@10.44.1.20:5060 Call-ID: 716c23be-d94f-060d-2f43-d9557f056...@10.44.1.20 CSeq: 41181 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 304 v=0 o=102 3079422269 3079422292 IN IP4 10.44.1.20 s=X-Lite c=IN IP4 10.44.1.20 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Attempting SIP protocol listen on: 10.44.1.20:5060 Established SIP protocol listen on: 10.44.1.20:5060 SEND TIME: 3079433234 SEND 0.0.0.100:5060 INVITE sip:100 SIP/2.0 Via: SIP/2.0/UDP 10.44.1.20:5060;rport;branch=z9hG4bK42C9872FBBB23E82A267FF9CA39FD454 From: informatica sip:1...@10.44.1.20;tag=93961341 To: sip:100 Contact: sip:1...@10.44.1.20:5060 Call-ID: 716c23be-d94f-060d-2f43-d9557f056...@10.44.1.20 CSeq: 41181 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 304 v=0 o=102 3079422269 3079422292 IN IP4 10.44.1.20 s=X-Lite c=IN IP4 10.44.1.20 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Attempting SIP protocol listen on: 10.44.1.20:5060 Established SIP protocol listen on: 10.44.1.20:5060 -- _ -- 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 failed: 408 timeout
SEND 0.0.0.100:5060 ?! -- _ -- 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 failed: 408 timeout
Hi, Please disable firewall and SElinux. 2010/7/9 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de SEND 0.0.0.100:5060 ?! -- _ -- 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] Call Failed
Robert La Ferla wrote: After so many rings when the party does not answer, my SIP phone says Call Failed. Why doesn't it just keep ringing? Here's the dial plan rule: exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],,r) exten = _NX,n,Hangup() Not that it's the cause, but get rid of the r. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Call Failed
After so many rings when the party does not answer, my SIP phone says Call Failed. Why doesn't it just keep ringing? Here's the dial plan rule: exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],,r) exten = _NX,n,Hangup() ___ --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] call failed: 499 Not acceptable here
Hi, Please help me through to overcome this error. I have my asterisk server and a xlite client in 172.16. network. I opened up port 5060 in my enterasyswireless router for my asterisk server and also opened port 8000 and 8001 for my xlite client. When i tried calling a remote system which successfullyconnected to my asterisk server i'm getting the error as " call failed: 499 Not acceptable here ". Thanks and Regards B.Srinivas Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call failed: 499 Not acceptable here
B.Srinivasa Rao wrote: Hi, Please help me through to overcome this error. I have my asterisk server and a xlite client in 172.16. network. I opened up port 5060 in my enterasys wireless router for my asterisk server and also opened port 8000 and 8001 for my xlite client. When i tried calling a remote system which successfully connected to my asterisk server i'm getting the error as call failed: 499 Not acceptable here . That reequently means that you have a codec problem. In sip.conf put disallow=all and allow=ulaw. If that works, you can experiment. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call failed to go through
Hi! I frequently get errors like Call failed to go through, reason X in /var/log/asterisk/messages Are the reasons explained anywhere? I did not find any info. Thanks, Carlos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call failed to go through, reason x
Hi! I frequently get errors like Call failed to go through, reason 0 in /var/log/asterisk/messages Are the reasons (0,3 and 5 in my case) explained anywhere? I did not find any info in the wiki. Thanks, Christian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
