[asterisk-bugs] [JIRA] (ASTERISK-28406) PJSIP compose invalid request-URI in ACK for re-INVITE

[Joshua C. Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28406?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=247089#comment-247089 ] 

Joshua C. Colp commented on ASTERISK-28406:
-------------------------------------------

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

As well Asterisk 15 is no longer supported, so any fixes or changes as a result of this will only go into supported branches. We will also need the configuration as there are options which can change the behavior.

> PJSIP compose invalid request-URI in ACK for re-INVITE 
> -------------------------------------------------------
>
>                 Key: ASTERISK-28406
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28406
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 15.7.2
>            Reporter: Dang Thanh Loi
>            Severity: Minor
>              Labels: pjsip
>
> I'm working with `moh_passthrough = yes` for asterisk lies between an UA and a Proxy loose routing.
> The call flow
> UA1 <-> Asterisk <-> Proxy (kamailio) <-> UA2
> and things go wrong for the call between Asterisk and UA2 (via Proxy) when receive re-INVITE hold from UA1
> Asterisk supposes to place the UA2's Contact in the R-URI header for subsequent in-dialog requests sent to the proxy. Asterisk does well for the ACK corresponding to the initial INVITE, but for subsequent re-INVITEs (and their ACKs), Asterisk placed the Proxy's Contact into the R-URI which make them not routable.



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