johnnybrian
IS-IT--Management
Hi Guys! I have a couple of sip lines, used mainly for twinning purposes. Problem is, when twinning using these trunks, the caller ID remaisn the number of the SIP trunk, it doesnt pass through who is calling.
Im definately doin something wrong, because its across multiple providers. Heres a trace:
589000957mS SIP Call Rx: 17
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;received=xx.xx.xx.xx;rport=5060;branch=z9hG4bK279f88f173b56e5824cefb4e1eb5ccc6
Record-Route: <sip:213.215.45.231:5080;lr=on>
Record-Route: <sip:213.215.45.230;lr=on>
From: "04861547xx" <sip:myname@ippi.fr>;tag=9a4d1e199e6f54d8
To: <sip:06149237xx@ippi.fr>;tag=gj-2k5-4cdaee90-00007461-0019a8b3Rdeb52263.b
Call-ID: c0a05753a92daa5e6f3779ebe6a492cf@xx.xx.xx.xx
CSeq: 1310624354 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:336149237xx@213.215.45.253;tnat=yes>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 21873 21873 IN IP4 213.215.45.245
s=session
c=IN IP4 213.215.45.245
t=0 0
m=audio 47854 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSuppff - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes
589002294mS SIP Call Rx: 17
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;received=xx.xx.xx.xx;rport=5060;branch=z9hG4bK279f88f173b56e5824cefb4e1eb5ccc6
Record-Route: <sip:213.215.45.231:5080;lr=on>
Record-Route: <sip:213.215.45.230;lr=on>
From: "04861547xx" <sip:itplan@ippi.fr>;tag=9a4d1e199e6f54d8
To: <sip:06149237xx@ippi.fr>;tag=gj-2k5-4cdaee90-00007461-0019a8b3Rdeb52263.b
Call-ID: c0a05753a92daa5e6f3779ebe6a492cf@xx.xx.xx.xx
CSeq: 1310624354 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:3361492373xx@213.215.45.253;tnat=yes>
Content-Length: 0
On the line, Send caller ID is set to Diversion Header, have tried the others as well.
I cant figure out why CALLER ID looks like this in the trace:
Call-ID: c0a05753a92daa5e6f3779ebe6a492cf@xx.xx.xx.xx
That cant be right, can it?
I look forward to your help.
Im on IPO500, 7.023
Im definately doin something wrong, because its across multiple providers. Heres a trace:
589000957mS SIP Call Rx: 17
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;received=xx.xx.xx.xx;rport=5060;branch=z9hG4bK279f88f173b56e5824cefb4e1eb5ccc6
Record-Route: <sip:213.215.45.231:5080;lr=on>
Record-Route: <sip:213.215.45.230;lr=on>
From: "04861547xx" <sip:myname@ippi.fr>;tag=9a4d1e199e6f54d8
To: <sip:06149237xx@ippi.fr>;tag=gj-2k5-4cdaee90-00007461-0019a8b3Rdeb52263.b
Call-ID: c0a05753a92daa5e6f3779ebe6a492cf@xx.xx.xx.xx
CSeq: 1310624354 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:336149237xx@213.215.45.253;tnat=yes>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 21873 21873 IN IP4 213.215.45.245
s=session
c=IN IP4 213.215.45.245
t=0 0
m=audio 47854 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSuppff - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes
589002294mS SIP Call Rx: 17
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;received=xx.xx.xx.xx;rport=5060;branch=z9hG4bK279f88f173b56e5824cefb4e1eb5ccc6
Record-Route: <sip:213.215.45.231:5080;lr=on>
Record-Route: <sip:213.215.45.230;lr=on>
From: "04861547xx" <sip:itplan@ippi.fr>;tag=9a4d1e199e6f54d8
To: <sip:06149237xx@ippi.fr>;tag=gj-2k5-4cdaee90-00007461-0019a8b3Rdeb52263.b
Call-ID: c0a05753a92daa5e6f3779ebe6a492cf@xx.xx.xx.xx
CSeq: 1310624354 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:3361492373xx@213.215.45.253;tnat=yes>
Content-Length: 0
On the line, Send caller ID is set to Diversion Header, have tried the others as well.
I cant figure out why CALLER ID looks like this in the trace:
Call-ID: c0a05753a92daa5e6f3779ebe6a492cf@xx.xx.xx.xx
That cant be right, can it?
I look forward to your help.
Im on IPO500, 7.023