IPO 9.1.12 Mobile twinning Issue

[karlzre]

Hi,

We have a customer using mobile twinning.

I found that when i call from outside to the number, the ipo try to twin with the mobile, but the call is rejected by the operator.

I found that the ipo send my number in the from field, so the operator didn't allow spoofing, he reject the call ( looks normal)
i can't find where iu have to force the ipo to send the user clip instead of the caller clip in FRom Field.

I've try to tick the "swap from and PAI/Diversion ( i can see in the sip trace that there is the correct number in the PAI)
didn't work.

Iv' set the Originator number in Forwarding and Twinning to the main number but it still send the caller CLIP.
I set the "send caller ID" to none, P Asserted ID, it didn't work too.
I've set the refresh metod to auto also, same issue.

I'm a little bit out of idea.

IP Call Tx: 30
                    INVITE sip:0491XXXXX@192.168.X.X;user=phone SIP/2.0
                    Via: SIP/2.0/UDP 192.168.X.X:5060;rport;branch=z9hG4bK03c7155b2f1d0635bba55719a8133657
                    From: "+32278XXXXX" <sip:+32278XXXXX@192.168.X.X;user=phone>;tag=7ed740ffc72f7dd6
                    To: <sip:0491XXXXX@192.168.X.X;user=phone>
                    Call-ID: 86cba6ca7fa98b08bd5f713880137097
                    CSeq: 734090918 INVITE
                    Contact: "+32278XXXX" <sip:+32278XXXX@192.168.X.X:5060;transport=udp>
                    Max-Forwards: 70
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY
                    Supported: timer,100rel
                    User-Agent: IP Office 9.1.12.0 build 212
                    P-Asserted-Identity: <sip:+3223XXXXX@192.168.X.X:5060>
                    Content-Type: application/sdp
                    Content-Length: 203
                    
                    v=0
                    o=UserA 745752734 4129903315 IN IP4 192.168.42.1
                    s=Session SDP
                    c=IN IP4 192.168.X.X
                    t=0 0
                    m=audio 49162 RTP/AVP 8 101
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-15


SIP Rx: UDP 192.168.X.X:5060 -> 192.168.X.X:5060
                    SIP/2.0 480 Temporarily Unavailable
                    Call-ID: 86cba6ca7fa98b08bd5f713880137097
                    CSeq: 734090918 INVITE
                    From: "+32278XXXX" <sip:+32278XXXX@192.168.X.X;user=phone>;tag=7ed740ffc72f7dd6
                    Reason: Q.850;cause=21;text="Call rejected"
                    To: <sip:0491XXXXX@192.168.X.X;user=phone>;tag=79F0
                    Via: SIP/2.0/UDP 192.168.X.X:5060;received=192.168.X.X;rport=5060;branch=z9hG4bK03c7155b2f1d0635bba55719a8133657
                    Content-Length: 0

ACIS - ACSS

 

[janni78]

Depends on the SIP provider but usually they look at Diversion Header do allow the call so try enabling that.

"Trying is the first step to failure..." - Homer

 

[karlzre]

we have found that in the uri it was set tuo use internal data but the trunk is not registred so that cause the issue.

We set it to Auto, and it's fixed now.

ACIS - ACSS