[asterisk-bugs] [JIRA] (ASTERISK-24569) user=phone is not added to From, Contact and Diversion header

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24569?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=223877#comment-223877 ] 

Matt Jordan commented on ASTERISK-24569:
----------------------------------------

The RFCs aren't super explicit about the usage of {{user=phone}}, and certainly don't mandate its usage. The fact that your provider is requiring it is ... odd.

That being said, after talking with a few other folks, if the user portion of a SIP URI is *known* to map to a PSTN DID, then you can certainly include {{user=phone}}.

I'm not sure how you're going to be able to make this determination for all of the fields you mention, but if you have a patch in mind, we can certainly entertain it. Re-opening.

> user=phone is not added to From, Contact and Diversion header
> -------------------------------------------------------------
>
>                 Key: ASTERISK-24569
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24569
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.14.1
>         Environment: cento 6.5
>            Reporter: Mark Petersen
>
> the problem is that user=phone is not added to the From, Contact and Diversion header, but is correctly added to the INVITE and To header
> This is a major problem as our Provider is switching to a new platform where they require these header, in order for os to set the outgoing CALLERID on our trunk
> [general]
> usereqphone=yes
> Set(CALLERID(name)=77777777);
> Set(CALLERID(num)=88888888);
> Set(CALLERID(ANI-num)=99999999);
> Set(CALLERID(rdnis)=66666666);
> Dial(SIP/55555555 at 192.168.0.2);
> INVITE sip:55555555 at 192.168.0.1;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1b298676;rport
> From: "77777777" <sip:88888888 at 192.168.0.1>;tag=as14ad576b
> To: <sip:55555555 at 192.168.0.2;user=phone>
> Contact: <sip:88888888 at 192.168.0.1:5060>
> Call-ID: 566480180a198f053ee9ba1016c0aef8 at 192.168.0.1
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Diversion: <sip:66666666 at 192.168.0.1>;reason=unknown



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