Caller id scrambled on SIP lines

[johnnybrian]

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

 

[johnnybrian]

really, noone? I just need the SIP lines to transfer the calling partys number via mobility, is that not possible?

 

[hairlessupportmonkey]

*Set Send Caller ID in the SIP trunk to "Diversion Header"

*Turn OFF Send Original calling party information for mobile twinning, in the System / Twinning TAB



ACSS - SME
General Geek

 

[johnnybrian]

Its done! Still no cigar

 

[hairlessupportmonkey]

who is the SIP provider?

ACSS - SME
General Geek

 

[redphone]

IP Office 7.0.23 has a problem with the Diversion header for Mobility calls. It sends the Original calling party caller ID in the Diversion header instead of the FROM header.

Try downgrading it to 7.0.5. This release send the original caller ID in the FROM header and twinned extension in the Diversion header.

The Call-ID header is not Caller ID. The Call-ID header uniquely identifies a particular invitation or all registrations of a particular client.

 

[johnnybrian]

Redphone; Thanks for clearing that up, so actually I should see the original callers number in the FROM header, in order for it to work?

The provider is french Ippi.fr but i seem to ahve the same problem with other providers! let me also show you my URI config, just to make sure that I havent done that wrong.

Hope there could be another way than downgrading!

 

[johnnybrian]

When I remove the CONTACT from my Uri and replace it with a * i get this when twinning (it shows 4852 as from, which is my local extension)and marks it as FAKE


SIP/2.0 403 Fake FROM - use From=id next time
Via: SIP/2.0/UDP 46.32.32.116:5060;rport=5060;branch=z9hG4bKb6b9c6b23ffa52a15fae0a1e6e1cb435
From: "Holger Axelgaar" <sip:4852@ippi.fr>;tag=63e568892dcf6e8a
To: <sip:0614923733@ippi.fr>;tag=a910c8153188470b2841623c513a131f.150a
Call-ID: 3196540751cadc55f1c7820ce7283eb6@46.32.32.116
CSeq: 1334097929 INVITE
Server: OpenSIPS (1.6.3-notls (i386/linux))
Content-Length: 0

 

[hairlessupportmonkey]

does the user have either a DDI or their DDI in their fields in the SIP tab on the user?

ACSS - SME
General Geek

 

[johnnybrian]

yes, it says
4852
My Name
4852

Should i check anonymous? I cant remove any of it it seems?

 

[hairlessupportmonkey]

just put the whole DDI in each of the fields!

ACSS - SME
General Geek

 

[Pepp77]

Have this working on my voiceflex trunks.

As per HSM I -
*Set Send Caller ID in the SIP trunk to "Diversion Header"
*Turn OFF Send Original calling party information for mobile twinning, in the System / Twinning TAB

The user then has their DDI number in the SIP tab on their user and the outgoing line ID for their mobile number is set to ensure the call uses the line group I have set up for Use User Data.

Turn on passthrough on the voiceflex portal and then when I call my DDI and the call forwards to my mobile I see the number of the caller and not the system.

You do need to ensure the provider allows this as I havent got it working on my Gamma IPDC trunks with the exact same setup.

| ACSS SME |

 

[johnnybrian]

Hm done all of that, but maybe its my provider who doesnt support diversion header...

Ill try to give them a call