[asterisk-bugs] [JIRA] (ASTERISK-29461) Contact header format for inbound calls. One way audio inbound calls behind NAT.

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29461?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=255074#comment-255074 ] 

Asterisk Team commented on ASTERISK-29461:
------------------------------------------

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

> Contact header format for inbound calls. One way audio inbound calls behind NAT. 
> ---------------------------------------------------------------------------------
>
>                 Key: ASTERISK-29461
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29461
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 18.3.0
>            Reporter: Maxim Grechikhin
>
> Dear community,
> I faced a situation of one way audio at inbound calls using PJSIP channel on Asterisk 18.3.0 installation. Previously I had the same situation when I was using SIP (chan_sip & sip.conf) channel and my SIP provider explained that they require Contact header in the following format: <sip:callee_number at external_ip:port> in transmitted portion of SIP headers to them. That time the issue was caused by private IP address used by Asterisk to form contact header and as I remember it has been fixed by "from_domain=" parameter in a trunk description. Here I would like to tell that despite the fact that Asterisk is located behind NAT, the firewall is configured correctly and allows inbound RTP sessions. Since migration to PJSIP channel the contact header looks like <sip: external_ip_addr:5060> and we do not have audio signal from remote parties. Is there a way to modify any settings in order to affect the format of contact header?  Or any other related suggestions much appreciated.



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