[asterisk-bugs] [JIRA] (ASTERISK-24691) Asterisk tries to transcode between g722 & h264

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24691?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224693#comment-224693 ] 

Rusty Newton commented on ASTERISK-24691:
-----------------------------------------

Why don't we see the following messages in your debug log

{noformat}
2015-01-15 09:29:05] WARNING[19055][C-0000007f]: channel.c:5070 ast_write: Codec mismatch on channel SIP/6032-0000008e setting write format to ulaw from slin16 native formats (g722|h264)
[2015-01-15 09:29:05] WARNING[19055][C-0000007f]: channel.c:5070 ast_write: Codec mismatch on channel SIP/6032-0000008e setting write format to slin16 from ulaw native formats (g722|h264)
{noformat}

Are you demonstrating two different scenarios, or did you exclude WARNING logger messages from your log?

Can you verify your debug log has all available log message types enabled (except dtmf,fax,security)?

> Asterisk tries to transcode between g722 & h264
> -----------------------------------------------
>
>                 Key: ASTERISK-24691
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24691
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling
>    Affects Versions: 13.1.0
>         Environment: CentOS 6.6
>            Reporter: Mark Farmer
>            Assignee: Matt Jordan
>         Attachments: myDebugLog.gz, sip.conf
>
>
> With the following codecs on SIP peers:
> disallow = all
> allow = g722,ulaw,alaw,h264
> When we try to originate a call via AMI or transfer a call that was answered via a queue, Asterisk seems to try to use h264 and transcode between g722 & h264.
> Console output:
> {noformat}
>   -- Called SIP/6059
>     -- Local/6059 at GageAgent-00000032;1 connected line has changed. Saving it until answer for SIP/ph-sip03-gn-lon1-0000008b
>     -- SIP/6059-0000008c is ringing
>     -- Local/6059 at GageAgent-00000032;1 is ringing
>     -- SIP/6059-0000008c answered Local/6059 at GageAgent-00000032;2
>     -- Local/6059 at GageAgent-00000032;1 answered SIP/ph-sip03-gn-lon1-0000008b
>     -- Channel Local/6059 at GageAgent-00000032;2 joined 'simple_bridge' basic-bridge <2765d56b-fcb8-46b5-8d83-7866feea3196>
>     -- Stopped music on hold on SIP/ph-sip03-gn-lon1-0000008b
>     -- Channel SIP/6059-0000008c joined 'simple_bridge' basic-bridge <2765d56b-fcb8-46b5-8d83-7866feea3196>
>     -- Channel SIP/ph-sip03-gn-lon1-0000008b joined 'simple_bridge' basic-bridge <70114587-b1d8-46a0-9f4e-ad77ed10da51>
>     -- Channel Local/6059 at GageAgent-00000032;1 joined 'simple_bridge' basic-bridge <70114587-b1d8-46a0-9f4e-ad77ed10da51>
>     -- SIP/6032-0000008e is ringing
>     -- SIP/6032-0000008e answered SIP/6059-0000008d
>     -- Channel SIP/6059-0000008d joined 'simple_bridge' basic-bridge <92d9bf20-c496-4b6c-86bf-2919395a56bf>
>     -- Channel SIP/6032-0000008e joined 'simple_bridge' basic-bridge <92d9bf20-c496-4b6c-86bf-2919395a56bf>
>     -- Channel Local/6059 at GageAgent-00000032;2 left 'simple_bridge' basic-bridge <2765d56b-fcb8-46b5-8d83-7866feea3196>
>     -- Channel Local/6059 at GageAgent-00000032;2 swapped with SIP/6059-0000008d into 'native_rtp' basic-bridge <92d9bf20-c496-4b6c-86bf-2919395a56bf>
>     -- Stopped music on hold on Local/6059 at GageAgent-00000032;2
>     -- Channel SIP/6059-0000008d left 'native_rtp' basic-bridge <92d9bf20-c496-4b6c-86bf-2919395a56bf>
>     -- Channel SIP/6059-0000008c left 'simple_bridge' basic-bridge <2765d56b-fcb8-46b5-8d83-7866feea3196>
>   == Spawn extension (DLPN_All, 6032, 50006) exited non-zero on 'SIP/6059-0000008d'
> [2015-01-15 09:29:05] WARNING[19055][C-0000007f]: channel.c:5070 ast_write: Codec mismatch on channel SIP/6032-0000008e setting write format to ulaw from slin16 native formats (g722|h264)
> [2015-01-15 09:29:05] WARNING[19055][C-0000007f]: channel.c:5070 ast_write: Codec mismatch on channel SIP/6032-0000008e setting write format to slin16 from ulaw native formats (g722|h264)
>     -- Registered SIP '6147' at xxx.xxx.xxx.xxx:61292
>     -- Registered SIP '6147' at xxx.xxx.xxx.xxx:8029
>     -- Channel SIP/ph-sip03-gn-lon1-0000008b left 'simple_bridge' basic-bridge <70114587-b1d8-46a0-9f4e-ad77ed10da51>
>     -- Channel Local/6059 at GageAgent-00000032;1 left 'simple_bridge' basic-bridge <70114587-b1d8-46a0-9f4e-ad77ed10da51>
>   == Spawn extension (voicemenu-custom-19, s, 5) exited non-zero on 'SIP/ph-sip03-gn-lon1-0000008b'
>     -- Channel Local/6059 at GageAgent-00000032;2 left 'simple_bridge' basic-bridge <92d9bf20-c496-4b6c-86bf-2919395a56bf>
>   == MixMonitor close filestream (mixed)
>   == End MixMonitor Recording SIP/ph-sip03-gn-lon1-0000008b
> {noformat}



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