On Tue, Jul 19, 2011 at 10:12 PM, Matthew J. Roth <[email protected]> wrote:
> Michael wrote: > > > > True. In the working system, LAN calls are also using G.729, while > > in the non-working system, LAN calls are in G.711 (supported but > > not prioritized by the phones) and only the SIP trunk to the ITSP > > is set to G.729. > > Can you set the phone to G.711 and try making a LAN call on the non- > working system. If a call that is G.711 from end-to-end doesn't have > the same problem it would be evidence of a codec translation issue. > That's the full trace of a call in G.711: Reliably Transmitting (no NAT) to 192.168.1.109:5060: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4751ec69;rport Max-Forwards: 70 From: "9000" <sip:[email protected]>;tag=as40d1788b To: <sip:[email protected]:5060> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.6.2.19) Date: Thu, 21 Jul 2011 07:12:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 261 v=0 o=root 1017783905 1017783905 IN IP4 192.168.1.10 s=Asterisk PBX 1.6.2.19 c=IN IP4 192.168.1.10 t=0 0 m=audio 17696 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 500 <--- SIP read from UDP:192.168.1.109:5060 ---> SIP/2.0 180 Ringing From: "9000"<sip:[email protected]>;tag=as40d1788b To: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe Call-ID: [email protected] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;branch=z9hG4bK4751ec69 Supported: replaces,100rel User-Agent: SIP Phone Contact: <sip:[email protected]:5060> Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/500-00000330 is ringing <--- SIP read from UDP:192.168.1.109:5060 ---> SIP/2.0 200 OK From: "9000"<sip:[email protected]>;tag=as40d1788b To: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe Call-ID: [email protected] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;branch=z9hG4bK4751ec69 Supported: replaces,100rel User-Agent: SIP Phone Contact: <sip:[email protected]:5060> Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 226 v=0 o=500 1096000000 0 IN IP4 192.168.1.109 s=Audio Session i=Audio Session c=IN IP4 192.168.1.109 t=0 0 m=audio 16384 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.109:16384 list_route: hop: <sip:[email protected]:5060> set_destination: Parsing <sip:[email protected]:5060> for address/port to send to set_destination: set destination to 192.168.1.109, port 5060 Transmitting (no NAT) to 192.168.1.109:5060: ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1265e92e;rport Max-Forwards: 70 From: "9000" <sip:[email protected]>;tag=as40d1788b To: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: FPBX-2.8.1(1.6.2.19) Content-Length: 0 --- -- SIP/500-00000330 answered SIP/Smile-0000032f [Jul 21 10:12:36] WARNING[13622]: dsp.c:1360 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 <--- SIP read from UDP:192.168.1.109:5060 ---> INVITE sip:[email protected] SIP/2.0 From: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe To: "9000"<sip:[email protected]>;tag=as40d1788b Call-ID: [email protected] CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-313-c04d6-373f24ae Max-Forwards: 70 Supported: replaces,100rel User-Agent: SIP Phone Contact: <sip:[email protected]:5060> Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 294 v=0 o=500 1096000001 0 IN IP4 192.168.1.109 s=SIPPhone Session i=Audio Session c=IN IP4 0.0.0.0 t=0 0 m=audio 16384 RTP/AVP 0 8 18 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (13 headers 14 lines) --- Sending to 192.168.1.109 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.109:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.109:5060 ;branch=z9hG4bK-313-c04d6-373f24ae;received=192.168.1.109 From: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe To: "9000"<sip:[email protected]>;tag=as40d1788b Call-ID: [email protected] CSeq: 1 INVITE Server: FPBX-2.8.1(1.6.2.19) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:[email protected]> Content-Length: 0 <------------> Audio is at 192.168.1.10 port 17696 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.1.109:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.109:5060 ;branch=z9hG4bK-313-c04d6-373f24ae;received=192.168.1.109 From: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe To: "9000"<sip:[email protected]>;tag=as40d1788b Call-ID: [email protected] CSeq: 1 INVITE Server: FPBX-2.8.1(1.6.2.19) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 237 v=0 o=root 1017783905 1017783905 IN IP4 192.168.1.10 s=Asterisk PBX 1.6.2.19 c=IN IP4 192.168.1.10 t=0 0 m=audio 17696 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.1.109:5060 ---> ACK sip:[email protected] SIP/2.0 From: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe To: "9000"<sip:[email protected]>;tag=as40d1788b Call-ID: [email protected] CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-313-c0521-2b8a87aa Max-Forwards: 70 User-Agent: SIP Phone Contact: <sip:[email protected]:5060> Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.1.109:5060 ---> INVITE sip:[email protected] SIP/2.0 From: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe To: "9000"<sip:[email protected]>;tag=as40d1788b Call-ID: [email protected] CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-321-c3cbc-39b225e8 Max-Forwards: 70 Supported: replaces,100rel User-Agent: SIP Phone Contact: <sip:[email protected]:5060> Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 300 v=0 o=500 1096000000 0 IN IP4 192.168.1.109 s=SIPPhone Session i=Audio Session c=IN IP4 192.168.1.109 t=0 0 m=audio 16384 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (13 headers 14 lines) --- Sending to 192.168.1.109 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.109:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.109:5060 ;branch=z9hG4bK-321-c3cbc-39b225e8;received=192.168.1.109 From: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe To: "9000"<sip:[email protected]>;tag=as40d1788b Call-ID: [email protected] CSeq: 2 INVITE Server: FPBX-2.8.1(1.6.2.19) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:[email protected]> Content-Length: 0 <------------> Audio is at 192.168.1.10 port 17696 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.1.109:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.109:5060 ;branch=z9hG4bK-321-c3cbc-39b225e8;received=192.168.1.109 From: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe To: "9000"<sip:[email protected]>;tag=as40d1788b Call-ID: [email protected] CSeq: 2 INVITE Server: FPBX-2.8.1(1.6.2.19) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 237 v=0 o=root 1017783905 1017783905 IN IP4 192.168.1.10 s=Asterisk PBX 1.6.2.19 c=IN IP4 192.168.1.10 t=0 0 m=audio 17696 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.1.109:5060 ---> ACK sip:[email protected] SIP/2.0 From: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe To: "9000"<sip:[email protected]>;tag=as40d1788b Call-ID: [email protected] CSeq: 2 ACK Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-322-c3ce9-34d41f51 Max-Forwards: 70 User-Agent: SIP Phone Contact: <sip:[email protected]:5060> Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.1.109:5060 ---> BYE sip:[email protected] SIP/2.0 From: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe To: "9000"<sip:[email protected]>;tag=as40d1788b Call-ID: [email protected] CSeq: 3 BYE Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-32b-c625a-41626b8 Max-Forwards: 70 Supported: replaces,100rel User-Agent: SIP Phone Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.1.109 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.109:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.109:5060 ;branch=z9hG4bK-32b-c625a-41626b8;received=192.168.1.109 From: <sip:[email protected]:5060 >;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe To: "9000"<sip:[email protected]>;tag=as40d1788b Call-ID: [email protected] CSeq: 3 BYE Server: FPBX-2.8.1(1.6.2.19) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> The same thing happens (or doesn't happen). MOH is not initiated when sendonly is received by Asterisk. I'm not an expert on this point, but I suspect that it's a system parameter somewhere and not related to codecs. > > > We tested with NAT set to "no" and "yes" and neither settings > > mattered. > > As long as the phone and the Asterisk server are both on the same LAN, > my recommendation would be to test with NAT set to "no". NAT is not > necessary unless there is a firewall between the phone and the > Asterisk server and setting it to "no" also eliminates a variable that > differentiates it from the working system. > We changed it to "no". It doesn't matter. > > > It should (have modules loaded for both formats). How do we check > > this? > > The following command should output a line for each module (as shown): > > # asterisk -rx 'module show' | egrep 'format_g729.so|format_pcm.so' > format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 5 > format_g729.so Raw G729 data 0 > That's what I get: [root@pbx ~]# asterisk -rx 'module show' | egrep 'format_g729.so|format_pcm.so' format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 format_g729.so Raw G729 data 0 What does the 5 (or in my case 0) stand for? Thanks.
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