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Summary:ASTERISK-16082: Asterisk randomly stops accepting calls in g729 format
Reporter:Richard Wilkinson (rickead2000)Labels:
Date Opened:2010-05-12 06:27:49Date Closed:2010-05-12 13:22:41
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Codecs/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) logger.txt
( 1) logger2.txt
( 2) translation.txt
Description:Asterisk will stop, seemingly at random, accepting calls in the g729 codec.  A restart will resolve this for a period of time.

Asterisk will not even try to call a sip peer that is only allowed to use g729.  If the sip peer is changed to allow any other codec (eg alaw) then this codec is used fine.

There are definitely enough licenses available as shown by "g729 show licenses"

****** ADDITIONAL INFORMATION ******

*CLI> g729 show licenses
0/0 encoders/decoders of 35 licensed channels are currently in use


The error posted the messages log is:-
[May 12 12:08:55] WARNING[12719] chan_sip.c: No audio format found to offer. Cancelling call to keybm-202

Entry in sip.conf is:-
[keybm-202]
type=friend
username=keybm-202
secret=abc276372
host=dynamic
qualify=yes
disallow=all
allow=g729
nat=yes
canreinvite=no
dtmfmode=rfc2833
mailbox=102@keybm
context=keybm-main
mohsuggest=keybm
limitonpeers=yes
call-limit=5
subscribecontext=keybm-hints
notifybusy=yes
limitonpeer=yes
notifyhold=yes


SIP Trace of a call being made to keybm-202:

U xx.xx.xx.xx:5060 -> yy.yy.yy.yy:5060
INVITE sip:+44161aaaaaa@yy.yy.yy.yy SIP/2.0.
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK02351abe;rport.
From: "+44161zzzzzz" <sip:+44161zzzzzz@domain.co.uk>;tag=as276908df.
To: <sip:+44161aaaaaa@yy.yy.yy.yy>.
Contact: <sip:+44161zzzzzz@xx.xx.xx.xx>.
Call-ID: 277708053b0e3b716196de15247667e4@domain.co.uk.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Date: Wed, 12 May 2010 11:21:08 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 289.
.
v=0.
o=root 14419 14419 IN IP4 xx.xx.xx.xx.
s=session.
c=IN IP4 xx.xx.xx.xx.
t=0 0.
m=audio 15238 RTP/AVP 0 3 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


U yy.yy.yy.yy:5060 -> xx.xx.xx.xx:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK02351abe;received=xx.xx.xx.xx;rport=5060.
From: "+44161zzzzzz" <sip:+44161zzzzzz@domain.co.uk>;tag=as276908df.
To: <sip:+44161aaaaaa@yy.yy.yy.yy>.
Call-ID: 277708053b0e3b716196de15247667e4@domain.co.uk.
CSeq: 102 INVITE.
Server: Asterisk PBX 1.6.2.7.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Contact: <sip:+44161aaaaaa@yy.yy.yy.yy>.
Content-Length: 0.
.


U yy.yy.yy.yy:5060 -> xx.xx.xx.xx:5060
SIP/2.0 503 Service Unavailable.
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK02351abe;received=xx.xx.xx.xx;rport=5060.
From: "+44161zzzzzz" <sip:+44161zzzzzz@domain.co.uk>;tag=as276908df.
To: <sip:+44161aaaaaa@yy.yy.yy.yy>;tag=as100290ed.
Call-ID: 277708053b0e3b716196de15247667e4@domain.co.uk.
CSeq: 102 INVITE.
Server: Asterisk PBX 1.6.2.7.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Length: 0.
Comments:By: Paul Belanger (pabelanger) 2010-05-12 08:37:08

We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs from an
Asterisk machine, for the purpose of helping bug marshals troubleshoot an issue:

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
By: Richard Wilkinson (rickead2000) 2010-05-12 10:48:42

logger.txt attached with relevant lines as per instructions
By: Paul Belanger (pabelanger) 2010-05-12 10:53:27

rickead2000: Please upload the complete trace of your call, there is some information we need prior to the stuff you posted.
By: Richard Wilkinson (rickead2000) 2010-05-12 11:14:04

logger2.txt uploaded.  Hopefully this is a full trace of the call - it's quite a busy production server and I don't want to clog the debug up with messages from other calls.
By: Richard Wilkinson (rickead2000) 2010-05-12 11:22:53

I have also uploaded the results from core show translation and it appears asterisk no longer thinks it can transcode g729
By: Paul Belanger (pabelanger) 2010-05-12 12:59:14

rickead2000: Just confirmed on #asterisk-dev, you will need to open a support ticket with Digium directly (support@digium.com).  Any fixes would have to me made in codec_g729.c, which is not included in Asterisk.
By: Paul Belanger (pabelanger) 2010-05-12 13:22:41

FYI: Tickets should be entered at http://www.digium.com/en/supportcenter/ not the support email address.