[asterisk-bugs] [JIRA] (ASTERISK-30377) Receiver and Caller can not hear the voice of each other

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30377?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=261071#comment-261071 ] 

Asterisk Team commented on ASTERISK-30377:
------------------------------------------

We appreciate the difficulties you are facing, however information request type issues 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.

If this issue is actually a bug please use the Bug issue type instead.

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

> Receiver and Caller can not hear the voice of each other 
> ---------------------------------------------------------
>
>                 Key: ASTERISK-30377
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30377
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 16.29.0
>            Reporter: usman Sajid
>
> Dear Asterisk Team
> I Hope every thing is fine with you. 
> I was using Asterisk 1.8.22.0 and configured more than 100 extensions and 4 to 5 SIP trunks some years ago. Suddenly I planned to upgrade my asterisk box. I configured the new asterisk server with Asterisk 16.29.0. every thing is working fine but my one sip trunk is registered but when incoming or outgoing calls land on this server both receiver and caller can not hear the voice of each other. 
> I investigate the RTP Packet and found that my asterisk server user new SSRC after 200 OK. 
> Below are the configuration details. 
> [general]
> context = default  
> allowguest = yes
> allowoverlap = yes 
> bindport=5060
> udpbindaddr = 0.0.0.0
> tcpenable = no  
> tcpbindaddr = 0.0.0.0
> transport = udp  
> srvlookup = yes  
> videosupport = yes
> localnet=10.5.0.0/255.255.0.0
> externip = 0.0.0.0
> qualify=yes
> nat=yes	
> subscribecontext = default
>  [PTCL]
> fromuser = XXXXXXXXXXX
> authname = XXXXXXXXXXX
> host = XX.XX.XX.XX
> type = peer
> nat = comedia
> dtmfmode=inband
> allow = ulaw
> allow = alaw
> qualify = yes
> directmedia=no
> context = UAN-Calling
> Kindly help me for the same and Thanks in advance. 



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