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

[Fazil (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24569?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226310#comment-226310 ] 

Fazil commented on ASTERISK-24569:
----------------------------------

Excellent its working fine

That provider also request for  with Media Connection information

These are sip debugs from their end server
----------------------------------------------------------------------------Sip debugs from their end by an out going call from our asterisk server-----------------------
 === Session Description Protocol ===
     Protocol Version                         : v=0
     Session Owner/Creator and session identi : o=root 377537064 377537064 IN IP4 10.100.20.10
     Session Name                             : s=Asterisk PBX 1.8.20.0
     Session Connection Information           : c=IN IP4 10.100.20.10
     Time the session is active               : t=0 0
     Media Name                               : m=audio 14830 RTP/AVP 8 101
       Media                                    : audio
       Port                                     : 14830
       Protocol type                            : RTP/AVP
       Media Format                             : 8 = PCMA
       Media Format                             : 101 = dynamic
     Media Attribute                          : a=rtpmap:8 PCMA/8000
     Media Attribute                          : a=rtpmap:101 telephone-event/8000
     Media Attribute                          : a=fmtp:101 0-16
     Media Attribute                          : a=ptime:20
     Media Attribute                          : a=sendrecv
---------------------------------------------------Sip debugs from their end by an out going call from an server what they exepect-----------------------
           Protocol Version                         : v=0
           Session Owner/Creator and session identi : o=CiscoSystemsSIP-GW-UserAgent 6070 8175 IN IP4 178.16.8.7
           Session Name                             : s=SIP Call
           Session Connection Information           : c=IN IP4 178.16.8.7
           Time the session is active               : t=0 0
           Media Name                               : m=audio 32706 RTP/AVP 8 101
             Media                                    : audio
             Port                                     : 32706
             Protocol type                            : RTP/AVP
             Media Format                             : 8 = PCMA
             Media Format                             : 101 = dynamic
           Media Connection information             : c=IN IP4 178.16.8.7         <- with Media Connection information
           Media Attribute                          : a=rtpmap:8 PCMA/8000
           Media Attribute                          : a=rtpmap:101 telephone-event/8000
           Media Attribute                          : a=fmtp:101 0-15
           Media Attribute                          : a=ptime:20

------------------------------------------------------------------------------------------------------------


Sip debug from our server as follows


NVITE sip:0553625772 at 172.16.84.6;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.20.10:5060;branch=z9hG4bK3b61bdfd
Max-Forwards: 70
From: <sip:576981000 at 10.100.20.10;user=phone>;tag=as2f333a3c
To: <sip:0553625772 at 172.16.84.6;user=phone>
Contact: <sip:576981000 at 10.100.20.10:5060>
Call-ID: 65161339488954b43193f8562f8ac7d4 at 10.100.20.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Wed, 21 Jan 2015 09:32:12 GMT
Session-Expires: 1800
Min-SE: 1800
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Allow-Events: telephone-event
Content-Disposition: session;handling=required
P-Asserted-Identity: <sip:576981000 at 10.100.20.10;user=phone>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 576210786 576210786 IN IP4 10.100.20.10
s=Asterisk PBX 1.8.20.0
c=IN IP4 10.100.20.10
t=0 0
m=audio 15996 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv



> 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
>         Attachments: chan_sip.24569.2.patch
>
>
> 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
> {noformat}
> [general]
> usereqphone=yes
> {noformat}
> {noformat}
> 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);
> {noformat}
> {noformat}
> 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
> {noformat}



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