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Summary:ASTERISK-30377: Receiver and Caller can not hear the voice of each other
Reporter:usman Sajid (usmansajid007)Labels:
Date Opened:2022-12-26 04:28:46.000-0600Date Closed:2022-12-26 04:28:49.000-0600
Priority:MinorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:16.29.0 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description: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.
Comments:By: Asterisk Team (asteriskteam) 2022-12-26 04:28:48.963-0600

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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

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].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/].

By: Asterisk Team (asteriskteam) 2022-12-26 04:28:49.651-0600

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