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Summary:ASTERISK-26347: Asterisk not sending out RTP packets in case of redirect call
Reporter:Denis S.Davydov (denis.davydov)Labels:
Date Opened:2016-09-08 03:47:36Date Closed:2016-09-08 12:28:42
Priority:MinorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:13.10.0-rc1 Frequency of
Occurrence
Related
Issues:
Environment:Virtual Machine on ESXi6 (VM Version 11, 4vCPU, 16Gb RAM), OS: CentOS release 6.8 (Final) x86_84Attachments:( 0) ukr8.cap
( 1) ukr8.txt
Description: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}
Comments:By: Asterisk Team (asteriskteam) 2016-09-08 03:47:37.424-0500

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].
By: Denis S.Davydov (denis.davydov) 2016-09-08 03:49:36.132-0500

ukr8.cap is tcpdump trace file
ukr8.txt is debug console (RTP & SIP for remote peer 62.221.34.22)
By: Rusty Newton (rnewton) 2016-09-08 12:28:32.857-0500

We appreciate the difficulties you are facing, however this does not appear to be a bug report and your request or comments would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

You might try the http://community.asterisk.org site first. If with the help of the community you determine in more detail what is happening and it appears to be a bug, then you can reopen this issue or open a new issue detailing the exact nature of what the bug is.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines


By: Rusty Newton (rnewton) 2016-09-08 12:30:00.455-0500

In addition, it is good to note that chan_sip is under extended support in 13 (community support) and if you need core support then you will want to use the new chan_pjsip channel driver which is where SIP development is currently directed.