[asterisk-bugs] [JIRA] (ASTERISK-27433) Call Monitor doesn't work with native bridge

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27433?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=244386#comment-244386 ] 

Joshua Colp commented on ASTERISK-27433:
----------------------------------------

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



> Call Monitor doesn't work with native bridge
> --------------------------------------------
>
>                 Key: ASTERISK-27433
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27433
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.18.2
>         Environment: Ubuntu 14.04
>            Reporter: Oguzhan Kayhan
>            Assignee: Unassigned
>            Severity: Minor
>         Attachments: 13.17-dial-output.txt, 13.18-dial-output.txt, messages, sip.conf, sip-debug.txt, sip-hsta.conf
>
>
> 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



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