[asterisk-bugs] [JIRA] (ASTERISK-24518) Avaya Asterisk sip trunk DTMF mis negotiated

[Jonathan White (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24518?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=223954#comment-223954 ] 

Jonathan White commented on ASTERISK-24518:
-------------------------------------------

Are you happy you have the right level of debug data coming in the log file now?
We tried inband, setting this both sides, Avaya and asterisk and we got the same results.
This sip configuration is via an Avaya Session Manager over udp but we also tried creating a sip trunk directly from Avaya Communication Manager over rcp, same results and finally a H.323 trunk direct from Avaya Communication Manager. So has little to do with call control and more to do with audio (expected) which is handeled in this instance by multiple G450 gateways.  
We found dialling in with X-Lite worked fine.

> Avaya Asterisk sip trunk DTMF mis negotiated
> --------------------------------------------
>
>                 Key: ASTERISK-24518
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24518
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_read
>    Affects Versions: 11.12.0
>         Environment: Centos 6.4
>            Reporter: Jonathan White
>            Assignee: Jonathan White
>         Attachments: capture201412061210.pcap, full
>
>
> I am upgrading from Asterisk ver 11.5.1 to 11.12.0
> An existing sip trunk is built from an Avaya Session Manager V6 to Asterisk
> rfc2833 is configured in sip.conf
> Asterisk V 11.5.1 negotiates and receives DTMF correctly (RTP type 101)
> Asterisk V 11.12.0 Does not receive DTMF but shows (RTP type 127)
> The Sip.conf and rtp.conf files are the same. There is no change to the Avaya configuration
> I have tried auto and inband in the sip.conf to see if I can match type 127 but this does not work either.



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