[asterisk-bugs] [JIRA] (ASTERISK-28878) chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16

[Joshua C. Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28878?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=250671#comment-250671 ] 

Joshua C. Colp commented on ASTERISK-28878:
-------------------------------------------

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

In this case the precise configuration, dialplan logic, SIP trace, and console log.

> chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
> ------------------------------------------------
>
>                 Key: ASTERISK-28878
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28878
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 16.10.0
>            Reporter: Joseph Ades
>            Severity: Minor
>
> I am migrating from asterisk 13 and have successfully utilized the pjsip_media_offer function. Very simple code, not much else to it, and worked for years.
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_MEDIA_OFFER
> After testing an asterisk 16 machine, using exactly the same code, I am unable to get it to work. The CLI reads the code, but does not do anything with it. Furthermore, after analyzing the packet capture, there is no evidence of the asterisk machine sending an INVITE to renegotiate the codec.
> I downgraded back to asterisk 13 and everything started to work again



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