Re: [asterisk-users] G.729 codec in pass-thru mode
Matthew, allow=all is defined in sip.conf for the ITSP's SIP peer. Additionally, ITSP supports g729 codec as we are able to send the traffic from other soft switch. In case g729 on asterisk box, as I mentioned earlier, call even doesn't go out of the asterisk box. Below extracts from log also indicate the same thing. [Jun 5 12:46:49] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456) [Jun 5 12:46:49] == Using SIP RTP CoS mark 5 [Jun 5 12:46:49] -- Couldn't call yyy.yyy.yyy.yyy/12127773456 [Jun 5 12:46:49] == Everyone is busy/congested at this time (0:0/0/0) Regards, Kamlesh Date: Tue, 4 Jun 2013 10:27:11 -0500 From: mr...@imminc.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Kamlesh Kumar wrote: SIP.conf [100] username=100 secret=password type=friend host=dynamic nat=yes canreinvite=no insecure=port disallow=all allow=ulaw allow=alaw allow=g729 context=asterisk qualify=no Is there also an allow=g729 line in sip.conf for the ITSP's SIP peer? SIP Trace: 201.xxx.xxx.xxx = SIP Softphone which originates the call xxx.xxx.xxx.xxx = Asterisk server yyy.yyy.yyy.yyy = ITSP ... --- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060 From: 100 sip:1...@xxx.xxx.xxx.xxx;tag=as643c20b1 To: sip:12127773...@yyy.yyy.yyy.yyy;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 102 INVITE Contact: sip:12127773...@yyy.yyy.yyy.yyy:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Content-Length: 234 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy s=SIP Media Capabilities c=IN IP4 zzz.zzz.zzz.zzz t=0 0 m=audio 21996 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - [Jun 3 13:11:31] --- (11 headers 11 lines) --- [Jun 3 13:11:31] Found RTP audio format 0 [Jun 3 13:11:31] Found RTP audio format 101 [Jun 3 13:11:31] Found audio description format PCMU for ID 0 [Jun 3 13:11:31] Found audio description format telephone-event for ID 101 [Jun 3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jun 3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996 [Jun 3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-34d9 is making progress passing it to SIP/100-34d8 [Jun 3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042 [Jun 3 13:11:31] Adding codec 0x4 (ulaw) to SDP [Jun 3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP This response from the ITSP says that only u-law may be used for the call. Please contact the ITSP and confirm that they actually support the G.729 codec. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] G.729 codec in pass-thru mode
Kamlesh Kumar wrote: allow=all is defined in sip.conf for the ITSP's SIP peer. Additionally, ITSP supports g729 codec as we are able to send the traffic from other soft switch. There must be some difference between your Asterisk servers. Please set them up for calling the ITSP with G.729 and provide the following CLI output from both of them. Be sure to preserve any differences when obscuring IP addresses and label the output clearly as G.729 working and G.729 fails. CLI core show version CLI sip show settings CLI sip show peer 100 CLI sip show peer ITSP's SIP peer CLI g729 show licenses Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] G.729 codec in pass-thru mode
Kamlesh Kumar wrote: SIP.conf [100] username=100 secret=password type=friend host=dynamic nat=yes canreinvite=no insecure=port disallow=all allow=ulaw allow=alaw allow=g729 context=asterisk qualify=no Is there also an allow=g729 line in sip.conf for the ITSP's SIP peer? SIP Trace: 201.xxx.xxx.xxx = SIP Softphone which originates the call xxx.xxx.xxx.xxx = Asterisk server yyy.yyy.yyy.yyy = ITSP ... --- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060 From: 100 sip:1...@xxx.xxx.xxx.xxx;tag=as643c20b1 To: sip:12127773...@yyy.yyy.yyy.yyy;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 102 INVITE Contact: sip:12127773...@yyy.yyy.yyy.yyy:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Content-Length: 234 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy s=SIP Media Capabilities c=IN IP4 zzz.zzz.zzz.zzz t=0 0 m=audio 21996 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - [Jun 3 13:11:31] --- (11 headers 11 lines) --- [Jun 3 13:11:31] Found RTP audio format 0 [Jun 3 13:11:31] Found RTP audio format 101 [Jun 3 13:11:31] Found audio description format PCMU for ID 0 [Jun 3 13:11:31] Found audio description format telephone-event for ID 101 [Jun 3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jun 3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996 [Jun 3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-34d9 is making progress passing it to SIP/100-34d8 [Jun 3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042 [Jun 3 13:11:31] Adding codec 0x4 (ulaw) to SDP [Jun 3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP This response from the ITSP says that only u-law may be used for the call. Please contact the ITSP and confirm that they actually support the G.729 codec. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] G.729 codec in pass-thru mode
1. Your softphone is not sending g729 [Jun 3 13:11:27] Capabilities: us - 0x10c (ulaw|alaw|g729), *peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing)*, combined - 0x4 (ulaw) I think free version of eyebeam doesn't come with g729, try Microsip or some other with g729 codec. If it is full version, check in the advanced sip settings and allow g729 2. canreinvite should be set to yes for using pass-thru mode check this interesting article Just FYI: Can we bypass Asterisk for RTP session?http://techyatwork.blogspot.ae/2010/10/can-asterisk-bypass-rtp-and-work-like.html 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] G.729 codec in pass-thru mode
Mark Henry wrote: 1. Your softphone is not sending g729 This was a SIP trace of a successful u-law call. In an earlier post Kamlesh provided a trace of a failed G.729 call which did not include the dialog between the Asterisk server and the ITSP. I asked for this trace so that I could see the codecs offered by the ITSP. 2. canreinvite should be set to yes for using pass-thru mode I believe that by pass-thru mode [1] Kamlesh means he wants to avoid transcoding from G.729 to another codec since that requires a license per channel. Pass- thru mode can be achieved with canreinvite=no as shown by the following line from the successful u-law SIP trace and Mark Michelson's asterisk-dev post: [Jun 3 13:11:32] -- Packet2Packet bridging SIP/100-34d8 and SIP/yyy.yyy.yyy.yyy-34d9 From [asterisk-dev] Native Bridging: terminology [2]: ...within SIP, native bridging has two subcategories. One, typically referred to as SIP native bridging is used when reINVITEs are enabled. The endpoints send their media directly to one another. The other subcategory is called Packet 2 Packet or P2P bridging. If reINVITEs are not enabled, but there are also no features that require the Asterisk core to be in the voice path, then the bridging will be done at the RTP layer of Asterisk. [1] http://www.voip-info.org/wiki/view/Asterisk+G.729+pass-thru [2] http://lists.digium.com/pipermail/asterisk-dev/2010-March/043053.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] G.729 codec in pass-thru mode
] --- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK78ec518c;rport=5060 From: 100 sip:1...@xxx.xxx.xxx.xxx;tag=as643c20b1 To: sip:12127773...@yyy.yyy.yyy.yyy;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 103 BYE Content-Length: 0 Regards, Kamlesh Date: Fri, 31 May 2013 08:50:38 -0500 From: mr...@imminc.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Kamlesh Kumar wrote: Yes that's correct, when I use u-law call works fine. In case of g729, I enabled sip debug with 'sip set debug on' and captured all the sip traces and got whatever I posted in last email. There was no other call on the system when I captured sip trace. Please suggest further proceedings. Kamlesh, Please provide a SIP trace (sip set debug on) of a successful u-law call. I'm especially interested in the dialog between the Asterisk server and the ITSP in this scenario. Also include the relevant sections of sip.conf and the dialplan. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] G.729 codec in pass-thru mode
Matthew, Yes that's correct, when I use u-law call works fine. In case of g729, I enabled sip debug with 'sip set debug on' and captured all the sip traces and got whatever I posted in last email. There was no other call on the system when I captured sip trace. Please suggest further proceedings. Regards, Kamlesh Date: Wed, 29 May 2013 08:42:39 -0500 From: mr...@imminc.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Kamlesh Kumar wrote: Call even doesn't go to the ITSP. I tried without AGI script and the results were same. Kamlesh, Your first message stated that the call is successful if the codec is u-law, so there must be communication between the Asterisk server and the ITSP. The key to understanding why the G.729 call fails is in this SIP signaling. How are you capturing the SIP trace? Are you enabling SIP debugging for the specific SIP softphone? If so, please use sip set debug on to enable it for all SIP packets. Then wait until there are no other calls on the Asterisk server, try another G.729 call, and post the CLI output. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] G.729 codec in pass-thru mode
Kamlesh Kumar wrote: Yes that's correct, when I use u-law call works fine. In case of g729, I enabled sip debug with 'sip set debug on' and captured all the sip traces and got whatever I posted in last email. There was no other call on the system when I captured sip trace. Please suggest further proceedings. Kamlesh, Please provide a SIP trace (sip set debug on) of a successful u-law call. I'm especially interested in the dialog between the Asterisk server and the ITSP in this scenario. Also include the relevant sections of sip.conf and the dialplan. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] G.729 codec in pass-thru mode
Hello Matthew, Call even doesn't go to the ITSP. I tried without AGI script and the results were same. Regards, Kamlesh Date: Tue, 28 May 2013 18:32:19 -0500 From: mr...@imminc.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Kamlesh, Please provide SIP traces of both call legs for a failed call. Your last message only included a SIP trace of the call leg from the SIP softphone to the Asterisk server. There was no SIP trace for the call leg from the Asterisk server to the ITSP and, as shown below, that is probably where the answer to your problem can be found. First, the call leg from the SIP softphone to the Asterisk server successfully negotiated G.729 as the codec: [May 28 11:51:34] Found RTP audio format 18 ... [May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) However, the call.php AGI script then failed to create the call leg from the Asterisk server to the ITSP: [May 28 11:51:34] -- Executing AGI(SIP/100-115f, call.php) [May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php [May 28 11:51:34] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456) [May 28 11:51:34] == Using SIP RTP CoS mark 5 [May 28 11:51:34] -- Couldn't call yyy.yyy.yyy.yyy/12127773456 [May 28 11:51:34] Scheduling destruction of SIP dialog '142182ef20750fda512f8d2b0b071...@xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE) [May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0) [May 28 11:51:34] -- SIP/100-115fAGI Script call.php completed, returning 0 [May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-115f' status is 'CHANUNAVAIL' Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] G.729 codec in pass-thru mode
Kamlesh Kumar wrote: Call even doesn't go to the ITSP. I tried without AGI script and the results were same. Kamlesh, Your first message stated that the call is successful if the codec is u-law, so there must be communication between the Asterisk server and the ITSP. The key to understanding why the G.729 call fails is in this SIP signaling. How are you capturing the SIP trace? Are you enabling SIP debugging for the specific SIP softphone? If so, please use sip set debug on to enable it for all SIP packets. Then wait until there are no other calls on the Asterisk server, try another G.729 call, and post the CLI output. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] G.729 codec in pass-thru mode
/12127773456 [May 28 11:51:34] Scheduling destruction of SIP dialog '142182ef20750fda512f8d2b0b071...@xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE) [May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0) [May 28 11:51:34] -- SIP/100-115fAGI Script call.php completed, returning 0 [May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-115f' status is 'CHANUNAVAIL' [May 28 11:51:34] --- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --- SIP/2.0 503 Service Unavailable v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 f: 100sip:1...@xxx.xxx.xxx.xxx;tag=c4446262 t: sip:12127773...@xxx.xxx.xxx.xxx;tag=as4e329d09 i: 052fcf17df558f7b CSeq: 2 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO k: replaces, timer l: 0 [May 28 11:51:34] --- SIP read from UDP:201.xxx.xxx.xxx:5060 --- ACK sip:12127773...@xxx.xxx.xxx.xxx SIP/2.0 To: sip:12127773...@xxx.xxx.xxx.xxx;tag=as4e329d09 From: 100sip:1...@xxx.xxx.xxx.xxx;tag=c4446262 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;rport Call-ID: 052fcf17df558f7b CSeq: 2 ACK Content-Length: 0 - [May 28 11:51:34] --- (7 headers 0 lines) --- [May 28 11:51:34] -- Executing AGI(SIP/100-115f, hangup.php) [May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.php [May 28 11:51:34] -- SIP/100-115fAGI Script hangup.php completed, returning 0 Thanks, Kamlesh From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Mon, 27 May 2013 11:51:53 -0400 Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Show us the sip debug for a failed call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kamlesh Kumar Sent: Monday, May 27, 2013 2:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] G.729 codec in pass-thru mode Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)--Asterisk(1.6.2.9)---SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456@default:1] AGI(SIP/100-, call.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php -- AGI Script Executing Application: (Dial) Options: (SIP/xxx.xxx.xxx.xxx/12127773456) -- Couldn't call xxx.xxx.xxx.xxx/12127773456 == Everyone is busy/congested at this time (0:0/0/0) -- SIP/100-AGI Script call.php completed, returning 0 -- Auto fallthrough, channel 'SIP/100-' status is 'CHANUNAVAIL' If I use, ulaw, call works fine. Thanks, Kamlesh -- _ -- 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] G.729 codec in pass-thru mode
Kamlesh, Please provide SIP traces of both call legs for a failed call. Your last message only included a SIP trace of the call leg from the SIP softphone to the Asterisk server. There was no SIP trace for the call leg from the Asterisk server to the ITSP and, as shown below, that is probably where the answer to your problem can be found. First, the call leg from the SIP softphone to the Asterisk server successfully negotiated G.729 as the codec: [May 28 11:51:34] Found RTP audio format 18 ... [May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) However, the call.php AGI script then failed to create the call leg from the Asterisk server to the ITSP: [May 28 11:51:34] -- Executing AGI(SIP/100-115f, call.php) [May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php [May 28 11:51:34] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456) [May 28 11:51:34] == Using SIP RTP CoS mark 5 [May 28 11:51:34] -- Couldn't call yyy.yyy.yyy.yyy/12127773456 [May 28 11:51:34] Scheduling destruction of SIP dialog '142182ef20750fda512f8d2b0b071...@xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE) [May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0) [May 28 11:51:34] -- SIP/100-115fAGI Script call.php completed, returning 0 [May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-115f' status is 'CHANUNAVAIL' Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] G.729 codec in pass-thru mode
Show us the sip debug for a failed call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kamlesh Kumar Sent: Monday, May 27, 2013 2:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] G.729 codec in pass-thru mode Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)--Asterisk(1.6.2.9)---SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456@default:1] AGI(SIP/100-, call.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php -- AGI Script Executing Application: (Dial) Options: (SIP/xxx.xxx.xxx.xxx/12127773456) -- Couldn't call xxx.xxx.xxx.xxx/12127773456 == Everyone is busy/congested at this time (0:0/0/0) -- SIP/100-AGI Script call.php completed, returning 0 -- Auto fallthrough, channel 'SIP/100-' status is 'CHANUNAVAIL' If I use, ulaw, call works fine. Thanks, Kamlesh -- _ -- 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
