[asterisk-bugs] [JIRA] (ASTERISK-26947) need ignoredspversion=1 for PJSIP

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26947?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236563#comment-236563 ] 

Asterisk Team commented on ASTERISK-26947:
------------------------------------------

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.

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].

> need ignoredspversion=1 for PJSIP
> ---------------------------------
>
>                 Key: ASTERISK-26947
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26947
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.15.0
>         Environment: Linux 64
>            Reporter: Private Name
>
> We need a setting
> ignoredspversion=1
> for PJSIP. This setting makes the old SIP channels work with Cisco and Broadsoft endpoints, which as of today are a huge part of the SIP trunking market in the US, namely: Verizon, Comcast, ATT, etc. PJSIP is not able to interop with those switches if they change media source after a connect. I had to go back and rewrite my app in old SIP.
> If you look in JIRA, there are many old cases where this is well documented. Years ago the developers fixed it for the old SIP channel but they forgot to do the same for the new PJSIP technology.



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