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

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24518?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=223951#comment-223951 ] 

Rusty Newton commented on ASTERISK-24518:
-----------------------------------------

So, I see the RTP and RFC2833 traffic in the pcap, but in I don't see the inbound RTP traffic in the RTP debug output from Asterisk.

I may be overlooking something, but it doesn't appear the RTP is getting into Asterisk. 

Besides DTMF, does inbound audio from Avaya to Asterisk work at all? That is, can you call into a Record and record a sound file, or call to another peer?



> 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: Rusty Newton
>         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.



--
This message was sent by Atlassian JIRA
(v6.2#6252)