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

Jarrod Sears (JIRA) noreply at issues.asterisk.org
Wed Feb 12 10:47:03 CST 2014


     [ https://issues.asterisk.org/jira/browse/ASTERISK-23251?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Jarrod Sears updated ASTERISK-23251:
------------------------------------

    Attachment: asterisk-db.sql
                nonrealtime-extensions.conf
                nonrealtime-sip.conf
                asterisk-g72960-problem-nonrealtime-log.txt
                asterisk-g72960-problem-nonrealtime-CLI.txt

CLI trace and log file for dialplan not using asterisk realtime. 

Realtime sip users were not used in the test and are not used in production. The database is only being used for the dialplan and CDRs
                
> 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
>            Assignee: Jarrod Sears
>            Severity: Critical
>         Attachments: asterisk-db.sql, asterisk-g72960-problem-nonrealtime-CLI.txt, asterisk-g72960-problem-nonrealtime-log.txt, g729_issue.log, nonrealtime-extensions.conf, nonrealtime-sip.conf, 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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