Summary: | ASTERISK-08122: Codec in SDP | ||
Reporter: | dario (dario) | Labels: | |
Date Opened: | 2006-11-13 05:27:31.000-0600 | Date Closed: | 2007-03-06 10:02:02.000-0600 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Codecs/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) debug.txt ( 1) SDP_log.txt ( 2) sip.conf | |
Description: | When I try make a connection from VoIP to PSTN, asterisk always put only one codec in SPD. It gets only first codec from globals section in sip.conf. Pleas find my logs: sip.conf and SDP_log.txt | ||
Comments: | By: Joshua C. Colp (jcolp) 2006-11-13 10:47:22.000-0600 This is a configuration issue. You are not using the peer entry in sip.conf to dial out, so it is using the global codec settings (which appear to be only g729). To fix this use the peer entry to dial out, and for further configuration issues please use an alternate support medium such as the asterisk-users mailing list. Thank you. By: dario (dario) 2006-11-14 02:02:09.000-0600 No it isn't. I attach my sip.conf. By: Joshua C. Colp (jcolp) 2006-11-15 14:48:25.000-0600 Reassigned to Anthony since he has a branch open now on fixing this issue. By: Anthony LaMantia (alamantia) 2006-11-15 17:15:30.000-0600 I have been workin gon a patch for this issue, i should be able to upload it later tonight or early tommrow for testing, By: Anthony LaMantia (alamantia) 2006-11-16 17:26:49.000-0600 the patch for this issue is not yet in working order. i will update you soon in the mean time you can check my branch on this in my team directory /asterisk/team/anthonyl/8350-codec-2 for updates as i commit them. when that happens i will also update this bug ticket By: Anthony LaMantia (alamantia) 2006-11-17 12:28:32.000-0600 Hi, i have added some logging to my branch for this bug located at http://svn.digium.com/svn/team/anthonyl/8350-codec-2/ can you please check it out and return the logging output with debugging turned on? By: dario (dario) 2006-11-21 01:54:32.000-0600 Hello, There is a log (debug.txt) Regards Darek By: Olle Johansson (oej) 2006-11-30 01:26:56.000-0600 Let's see 1. The INVITE from Asterisk has ilbc and alaw 2. Cisco responds with only ALAW - We have an ALAW call I don't see the problem you are reporting here - Asterisk has several codecs in invites and 200 OK. alamantia and file: where do you see an error? I might be missing something... By: dario (dario) 2006-11-30 08:13:42.000-0600 Hello, Could you explain me why asterisk send INVITE without g729 codec? Regards darek By: Olle Johansson (oej) 2006-11-30 10:47:05.000-0600 Can you please copy that INVITE. The asterisk invite in debug.txt is this, which includes alaw and ilbc. INVITE sip:8344@10.10.119.225 SIP/2.0 Via: SIP/2.0/UDP 10.10.119.251:5060;branch=z9hG4bK3aef7a86;rport Max-Forwards: 70 From: "9971" <sip:9971@10.10.119.251>;tag=as0ce8b33f To: <sip:8344@10.10.119.225> Contact: <sip:9971@10.10.119.251> Call-ID: 412f6751271edc7931c93b4437801a99@10.10.119.251 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 20 Nov 2006 21:33:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 289 v=0 o=root 2483 2483 IN IP4 10.10.119.248 s=session c=IN IP4 10.10.119.248 t=0 0 m=audio 16442 RTP/AVP 8 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv By: Anthony LaMantia (alamantia) 2006-12-06 17:28:38.000-0600 hi dario, i have updated my branch with one more line of debugging, can you test it with this one and paste the results, this should be the last bit of debugging information that i will need to continue work on this issue. By: Olle Johansson (oej) 2006-12-07 02:03:16.000-0600 alamantia: What's the issue??? By: Anthony LaMantia (alamantia) 2006-12-07 04:13:02.000-0600 it seems the sdp from the invite you pasted above should include a mapping for g729 By: Olle Johansson (oej) 2007-02-01 15:30:00.000-0600 dario: Do you have g.729 licenses installed in your asterisk? By: dario (dario) 2007-02-05 09:28:56.000-0600 Yes, I have. By: Serge Vecher (serge-v) 2007-02-06 12:35:38.000-0600 alright, this bug seems to be going nowhere ... Let's see, the problem is that "it gets only first codec from globals section in sip.conf". Ok, let's look at the respective entries from you sip.conf. We have "allow=g729,alaw". Now, I'm not sure if that's a supported way of allowing codecs. If you look in sip.conf.sample, you should be using: allow=g729 allow=alaw By: Brian West (bkw918) 2007-02-06 13:06:29.000-0600 yes allow=codec,codec,codec is supported. /b By: Serge Vecher (serge-v) 2007-02-21 15:07:40.000-0600 dario, can you please include a new debug log from the latest (r > 55900) 1.4? By: Serge Vecher (serge-v) 2007-03-06 10:02:01.000-0600 no response from reporter; if still an issue with 1.4.1, please reopen the report with debugging information as requested. |