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 03:47:36 | Date Closed: | 2016-09-08 12:28:42 |
Priority: | Minor | Regression? | |
Status: | Closed/Complete | Components: | 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_84 | Attachments: | ( 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. |