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

[Mark Petersen (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24569?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Mark Petersen updated ASTERISK-24569:
-------------------------------------

    Description: 
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


  was:
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
Max-Forwards: 70
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
User-Agent: ipnordic
Date: Mon, 01 Dec 2014 08:33:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Diversion: <sip:66666666 at 192.168.0.1>;reason=unknown
Content-Type: application/sdp
Content-Length: 240


> 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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