[asterisk-bugs] [JIRA] (ASTERISK-23251) G729:60 packet size not being set correctly in SIP Invite

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23251?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=215091#comment-215091 ] 

Rusty Newton commented on ASTERISK-23251:
-----------------------------------------

Can't reproduce this in a non-realtime environment, using your sip.conf with an added extension like:

{noformat}
[6001]
type=friend
host=dynamic
secret=6001
context=from-internal
{noformat}

"sip show peer 6001" shows the peers codec as set to allow "g729:60" and that is what I see go out in INVITE messages and 200 OKs.

Either this is a misconfiguration, or maybe the issue only occurs with a realtime configuration.

Can you provide:

 * a database dump of the sip peer experiencing the issue? Also the output of "sip show peer <peername>" for the peer?
 * any /etc/asterisk config files relevant to your realtime configuration
 * Your sip peers table schema

You might also want to test for yourself, the same scenario outside of a realtime configuration to make sure I didn't miss something.
                
> G729:60 packet size not being set correctly in SIP Invite
> ---------------------------------------------------------
>
>                 Key: ASTERISK-23251
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23251
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling, Codecs/General, Core/CodecInterface
>    Affects Versions: 1.8.11.1, 11.7.0
>         Environment: CentOS release 6.5 (Final) x64
> MySQL Database
> Dell R320 Server and Dell R210II Server
>            Reporter: Jarrod Sears
>            Severity: Critical
>         Attachments: g729_issue.log, sip.conf
>
>
> Incoming calls to the server correctly negotiate to g729:60.
> The sip.conf is set to:
> disallow=all
> allow=g729:60
> The outbound leg of the server then sends a SIP invite out specifying g729:20.
> I've tried setting up specific SIP peers and also using the general codec settings, both experience the same issue.
> I have also tried using the SET(SIP_CODEC=G729:60) command, which does not work.
> I have tried using the a server with Digium's g729 codec. I've also tried on a server without a g729 codec. Both experience the same issue.
> I originally experienced the issue on the 1.8.11.1 (Asterisk 1.8.11-cert1) version that we are using generally for production. I recently installed a new copy of Asterisk 11.7.0 and have the same problem.

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