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Summary:ASTERISK-27433: Call Monitor doesn't work with native bridge
Reporter:Oguzhan Kayhan (gobris)Labels:
Date Opened:2017-11-20 08:34:09.000-0600Date Closed:
Priority:MinorRegression?
Status:Open/NewComponents:Channels/chan_sip/General
Versions:13.18.2 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Ubuntu 14.04Attachments:( 0) 13.17-dial-output.txt
( 1) 13.18-dial-output.txt
( 2) messages
( 3) sip.conf
( 4) sip-debug.txt
( 5) sip-hsta.conf
Description:Hello
I am trying to record the conversations via AMI interface.
On my previous version (11.5.0) when i set the monitor command it was recording fine.

My sip.conf has the following config
{code}
[panel_number](!)
context=CommPanels
type=friend
host=dynamic
secret=xxxx
directmedia=no
canreinvite=no
callgroup=1
pickupgroup=1
;disallow=all
allow=all
dtmfmode=auto
nat=force_rport,comedia
{code}

and i have 2 users with this config.
But when i dial eachother..
I have the following

{code}
   -- Called SIP/4511
   -- SIP/4511-00000001 is ringing
      > 0x7f1e2c00e5c0 -- Strict RTP learning after remote address set to: 78.189.8.164:8000
   -- SIP/4511-00000001 answered SIP/4510-00000000
   -- Channel SIP/4511-00000001 joined 'simple_bridge' basic-bridge <ddd40dcc-439d-40b1-8637-ad0686e7b166>
   -- Channel SIP/4510-00000000 joined 'simple_bridge' basic-bridge <ddd40dcc-439d-40b1-8637-ad0686e7b166>
      > Bridge ddd40dcc-439d-40b1-8637-ad0686e7b166: switching from simple_bridge technology to native_rtp
      > Locally RTP bridged 'SIP/4510-00000000' and 'SIP/4511-00000001' in stack
{code}

And after this, if i send ami monitor command it says success but records an 44 bytes of wav file instead of conversation.


ps: tried with directmedia=yes same result
Comments:By: Asterisk Team (asteriskteam) 2017-11-20 08:34:10.708-0600

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: Richard Mudgett (rmudgett) 2017-11-20 09:42:14.210-0600

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

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

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

We need a log containing the complete call that also includes the SIP signaling (sip set debug on) and AMI debugging (manager set debug on).  After collecting the log attach it to the issue with a .txt extension.
By: Oguzhan Kayhan (gobris) 2017-11-21 03:11:48.716-0600

I think I found what was the cause..
We had a problem on using pjsua clients to call eachother with slin codec during 11.5.0

And as a workaround we were using a modification on rtp_engine.c file.
Chaning the payloads of slin
add_static_payload(10, ast_format_slin, 0); /* 2 channels */
       add_static_payload(11, ast_format_slin, 0); /* 1 channel */

as
add_static_payload(121, ast_format_slin, 0); /* 2 channels */
add_static_payload(120, ast_format_slin, 0); /* 1 channel */

and it was working fine with 11.5.0  but during v13 upgrade this change caused instability (dont know why)

By: Oguzhan Kayhan (gobris) 2017-11-21 03:17:40.475-0600

I am adding the two outputs. 13.17 was without the mod on rtp_engine.c  13.18 with mod.


By: Oguzhan Kayhan (gobris) 2017-11-21 04:53:51.121-0600

Another update..
No that wasnt the cause.
I tried several things..
directmedia;= yes/no    adding w or W to Dial parameters..
Nothing worked.. Everytime my calls connected native_rtp whatever i do

By: Oguzhan Kayhan (gobris) 2017-11-21 06:32:49.311-0600

Workaround that i did for now is remove native_bridge module and everything started to work as expected.
Right now i can monitor the calls from AMI interface

By: Benjamin Keith Ford (bford) 2017-12-04 16:20:16.884-0600

Thanks for reporting this. I was able to reproduce it and opened up an issue for it.
By: Oguzhan Kayhan (gobris) 2018-08-03 04:16:39.932-0500

Any updates on the following issue?

By: Joshua C. Colp (jcolp) 2018-08-03 04:23:03.176-0500

Your issue is in the queue. Your patience is appreciated as a developer may work the issue when time and resources become available.

Asterisk is an open source project and community members work the issues on a voluntary basis. You are welcome to develop your own patches and submit them to the project.[1]

If you are not a programmer and you are in a hurry to see a patch provided then you might try rallying support on the Asterisk users mailing list or forums.[2] Another alternative is offering a bug bounty on the asterisk-dev mailing list.[3] Often a little incentive can go a long way.

[1]: https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
[2]: http://www.asterisk.org/community/discuss
[3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties