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

Matt Jordan (JIRA) noreply at issues.asterisk.org
Thu Jan 15 19:43:34 CST 2015


     [ https://issues.asterisk.org/jira/browse/ASTERISK-24691?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan updated ASTERISK-24691:
-----------------------------------

    Assignee: Mark Farmer
      Status: Waiting for Feedback  (was: Triage)

> 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: Mark Farmer
>
> 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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