[asterisk-bugs] [JIRA] (ASTERISK-25428) Codec Negotation failed in transfer Szenario between two Server

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25428?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227690#comment-227690 ] 

Asterisk Team commented on ASTERISK-25428:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> Codec Negotation failed in transfer Szenario between two Server
> ---------------------------------------------------------------
>
>                 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
>
> 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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