[asterisk-bugs] [JIRA] (ASTERISK-26670) [patch] Outgoing SIP-URI Dialing via PJSIP

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26670?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=253343#comment-253343 ] 

Asterisk Team commented on ASTERISK-26670:
------------------------------------------

This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable.

> [patch] Outgoing SIP-URI Dialing via PJSIP
> ------------------------------------------
>
>                 Key: ASTERISK-26670
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26670
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_session
>    Affects Versions: 13.13.1, 14.2.1
>            Reporter: Alexander Traud
>      Target Release: 13.14.0, 14.3.0, 15.0.0
>
>         Attachments: pjsip_dialplan_sipdomain.patch
>
>
> Asterisk can be used as [B2BUA|//en.wikipedia.org/wiki/Back-to-back_user_agent]. In that scenario, Asterisk is able to cope/fix/overcome software bugs in VoIP/SIP phones. Therefore, some VoIP/SIP providers use their Asterisk not only as Registrar but as Outbound Proxy – for all outgoing calls. Furthermore, some VoIP/SIP apps use the Registrar as Outbound Proxy automatically, because they cannot dial other SIP proxies themselves. In these scenarios, Asterisk must be able to detect whether an extension can be resolved internally or whether a remote domain must be contacted (Outgoing SIP-URI Dialing).
> For example, the user dials <bob at example.com> on his phone. Then, Asterisk should not dial <sip:bob> but the whole URI including the domain: <sip:bob at example.com>. For this, Asterisk must know that the domain is not the local domain (otherwise, a local "bob" is called) and Asterisk must know the remote domain to start contacting its SIP proxy.
> In the channel driver chan_sip, this was solved via the dialplan parameter SIPDOMAIN. Currently with Asterisk 13.13, I am not aware of a similar approach for the new SIP channel driver res_pjsip, which is based on PJSIP. The same rules and extensions for chan_sip should work for res_pjsip as well. Therefore, this issue report is about implementing SIPDOMAIN in res_pjsip.
> The attached patch makes sure, SIPDOMAIN is set even when res_pjsip is used. PJSIP_HEADER(read, To) did not work for me, because I do not need the URI in the header To but the Request-URI actually.



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