[asterisk-bugs] [JIRA] (ASTERISK-25428) Codec negotation fails 'No compatible codecs, not accepting this offer!' in transfer scenario between two servers when using SILK and SPEEX

Rusty Newton (JIRA) noreply at issues.asterisk.org
Fri Oct 16 17:41:32 CDT 2015


     [ https://issues.asterisk.org/jira/browse/ASTERISK-25428?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-25428:
------------------------------------

    Summary: Codec negotation fails 'No compatible codecs, not accepting this offer!' in transfer scenario between two servers when using SILK and SPEEX  (was: Codec negotation fails in transfer scenario between two servers)

> Codec negotation fails 'No compatible codecs, not accepting this offer!' in transfer scenario between two servers when using SILK and SPEEX
> -------------------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-25428
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25428
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling
>    Affects Versions: 11.19.0
>         Environment: ubuntu precise and trusty
>            Reporter: Peter Katzmann
>            Assignee: Unassigned
>         Attachments: codec_neg_prob.pcap, servear_a.log, servear_b.log, servera.log, servera_sip.conf, servera.sip.conf, serverb.log, serverb_sip.conf, serverb.sip.conf, vmfail2.pcap
>
>
> There are two Servers (A and B) connected over wan via silk and speex
> There is user 7000 on Server A and user 8000 on Server B.
> Voicemail will only be handled from server A.
> User 8000 has forwarding to voicemail after 5 Seks.
> Now User 7000 calls user 8000, when User 8000 picks up teh phone everything is OK.
> If the call is expired it will be forwarded to the voicemail server A, in this case the call fails with no acceptable codec offer.
> When i modify the sip conf for server A or Server B to alow ulaw/alaw or g722 voicemail works as expected.



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