[asterisk-bugs] [JIRA] (ASTERISK-26347) Asterisk not sending out RTP packets in case of redirect call

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26347?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232212#comment-232212 ] 

Asterisk Team commented on ASTERISK-26347:
------------------------------------------

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].

> Asterisk not sending out RTP packets in case of redirect call
> -------------------------------------------------------------
>
>                 Key: ASTERISK-26347
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26347
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.10.0-rc1
>         Environment: Virtual Machine on ESXi6 (VM Version 11, 4vCPU, 16Gb RAM), OS: CentOS release 6.8 (Final) x86_84
>            Reporter: Denis S.Davydov
>            Severity: Minor
>
> I have Asterisk 13.10.0-rc1 inside my network with private address, also I have bidirectional (1:1) nat mapping from 212.65.93.74 to this address. My ouside provider is 62.221.34.22.
> Calls from outside to my Asterisk is working fine! I have RTP flow. The same thing for calls to outside from one of my internal phones connected to my Asterisk within private addresses. Everything works fine. But if I get the call from outside and redirect it by Dial app back to provider on another callee, I saw no any RTP traffic via Asterisk. Could you tell me why?
> Scheme: A calls B, B calls C
> (A - external phone from my SIP provider, B - the extension in my Asterisk, C - another external phone I call via my SIP provider).
> See attachments. Output debug information about calls and also dump file.
> From sip.conf:
> {code}
> externip=212.65.93.74
> localnet=192.168.0.0/255.255.0.0
> ...
> [vega]
> type=peer
> trunkname=vega
> host=62.221.34.22
> context=from-trunk
> insecure=invite
> disallow=all
> allow = alaw
> nat = no
> directmedia = no
> dtmfmode = rfc2833
> qualify=yes
> {code}



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