Passing on Remote Caller ID in SIP 3

[HarryNorstarGuy]

Hello

I am having a problem configuring something I thought would be very basic.

this is on an IP O 7.0 latest release (Q4 '11) customer is using SIP trunks from Coredial. we need to make sure that in a call flow where an incoming calls get forwarded to an outside number the original caller ID should go along with the call. this was a simple setting for a PRI but with SIP it doesn't work. well according to the book it would work but in reality it doesn't.

the sample call flow is as follows:
Remote Callers ID:8454922020 call into IPO on Trunk 18, dials extension 665, which is unconditionally forwarded to 718452225960,
shortcode 7 dials out on trunk 18, (which has REFER on on Incoming and outgoing) the calls goes out on trunk 18 but I CANNOT GET the CID (or the "FROM" part of the header to be "8454922020"

any help on this would be greatly appreciated.

Config file available upon request.

 

[tlpeter]

You must have the Q3 release because the Q4 is about to come on the end of the month

This should work when you are allow to do this.
You could try to check the option for twinning in the system settings.
But i am not sure if this will help on unconditional forwarding.


BAZINGA!

I'm not insane, my mother had me tested!

 

[HarryNorstarGuy]

Yea - its Q3,

believe me, I have changed around all the settings to no avail. I changed the Twinning Setting (System Twinning.

 

[HarryNorstarGuy]

What bothers me even more is that this same setting used to work. but now it doesn't - its mind buggeling!

 

[tlpeter]

Did you try changing the "call routing method"?


BAZINGA!

I'm not insane, my mother had me tested!

 

[HarryNorstarGuy]

I have not as "Call Routing Method" is used to set what info the IPO is using to route incoming calls but has not affect on how Outgoing calls are routed. But if you feel this can make the difference I shall try and let you know.

 

[tlpeter]

I remember when i changed this once that it sended me the extension number to my mobile phone so it did change the CLI that was send out.


BAZINGA!

I'm not insane, my mother had me tested!

 

[HarryNorstarGuy]

it does send the extension to the mobile that's the problem, I am trying to get the original CID!

here is how all the settings are on the SIP trunk:
Send Caller ID: P_ASSERTED_ID
Call Routing Method: REQUEST URI
Sip URI Settings:
Local URI:*
Contact: *
Display Name: *
PAI: None

...

here is how the invite message from the IP O looks like:

------------------------------------------------------------------
INVITE sip:18452225960@sip.expediatel.com SIP/2.0
Via: SIP/2.0/UDP 96.56.128.114:5060;rport;branch=z9hG4bKac7150e1bdf1db92f8ae725ab72b22a2
From: "8454922020" <sip:665@sip.expediatel.com>;tag=1b27eab921d7a78f
To: <sip:18452225960@sip.expediatel.com>
Call-ID: 9d7fd25fa547fb138d14e0ede87f3203@96.56.128.114
CSeq: 1196391311 INVITE
Contact: "8454922020" <sip:665@96.56.128.114:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
P-Asserted-Identity: "8454922020" <sip:665@96.56.128.114:5060>
Content-Length: 300
------------------------------------------------------------------
Note the "8454922020" is the CID of the original call, it is present in the following headers:
From
Contact
P-Asserted-indentity

The problem is that its not part of the URI, in other words, if the P-asserted-id header would of look like this:

P-Asserted-Identity: "8454922020" <sip:8454922020@96.56.128.114:5060>

I would of gotten the desired results.

to sum it up, the IP Office is trying to do something (as is evidence from the fact that the original number 8454922020 appears in the INVITE in 3 different headers) but its not doing the right thing. and therefore I am up 2 am tackeling this issue!

 

[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

 

[HarryNorstarGuy]

Support Monkey, I tried that, I still get the same results. Here is how the resulting Invite looks like:

------------------------------------------------------------------
INVITE sip:18452225960@sip.expediatel.com SIP/2.0
Via: SIP/2.0/UDP 96.56.128.114:5060;rport;branch=z9hG4bKa803fe3d1ede3600320c7ffdc5e7a282
From: "8454922020" <sip:665@sip.expediatel.com>;tag=a0ef7ea1d4cc1986
To: <sip:18452225960@sip.expediatel.com>
Call-ID: 9a33f797d9f23ef735ad4682d2bb273d@96.56.128.114
CSeq: 2120610201 INVITE
Contact: "8454922020" <sip:665@96.56.128.114:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Diversion: "Harry" <sip:665@96.56.128.114:5060>;reason=unconditional
Content-Length: 301
------------------------------------------------------------------

the SIP URI still has 665 to the left of the @

any help on this would be greatly appreciated.

 

[pkdev]

Maybe you should fill the SIP tab of the user with the number you want to be sent ?

 

[HarryNorstarGuy]

PKDEV. This will not help. I want the CID number of the firstincoming call to be sent.

Setting a number is the SIP is fine for regular outgoing calls. I use a shortcode that access a line group that in the URI has the setting: USE INTERNAL DATA and the data in the SIP Tab is used. It works just fine!

To me its a bug in the IP Office. It does not work as described in its own documentation.

 

[hairlessupportmonkey]

It works 100% better with Mobile Twinning.

if licence not available to you, try putting in the forwarded users short code ? /.SS / Dial / LG (or try SS.)

ACSS - SME
General Geek

 

[DennyHeack]

Stupid question maybe, but does your SIP provider support Diversion/Deflection?

 

[HarryNorstarGuy]

Denny - the provider does support it.

Hairless support monkey - neither adding the short-code, nor using twinning helps.

here is how the invite looks after using twinning and turning "Send Original Party Information for Mobile Twinning:" is turned ON.
INVITE sip:18452225960@sip.expediatel.com SIP/2.0
Via: SIP/2.0/UDP 96.56.128.114:5060;rport;branch=z9hG4bKb29d8a15a0f7b944fc6f3531746b9cdb
From: "8454922020" <sip:665@sip.expediatel.com>;tag=312e28647f160e86
To: <sip:18452225960@sip.expediatel.com>
Call-ID: d956572927680bdf7e86fa3dbdccb607@96.56.128.114
CSeq: 995646056 INVITE
Contact: "8454922020" <sip:665@96.56.128.114:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Diversion: "Harry" <sip:665@96.56.128.114:5060>;reason=unconditional
Content-Length: 300


here is how the invite looks like with the added short-code using unconditional forwarding:

INVITE sip:18452225960@sip.expediatel.com SIP/2.0
Via: SIP/2.0/UDP 96.56.128.114:5060;rport;branch=z9hG4bK14ee423d473239da80cf480fced97dbf
From: "Harry" <sip:665@sip.expediatel.com>;tag=72b8459f0eded2c4
To: <sip:18452225960@sip.expediatel.com>
Call-ID: bbda2874fb25866421073ee048c82b14@96.56.128.114
CSeq: 986670249 INVITE
Contact: "Harry" <sip:665@96.56.128.114:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
P-Asserted-Identity: "8454922020" <sip:665@96.56.128.114:5060>
Content-Length: 300


I also noticed the following line in the help screen:
"User and User Rights short codes are only applied to numbers dialed by that user. For example they are not applied to calls forwarded via the user."

any help further would be greatly appreciated.

Thanks

 

[redphone]

The easiest fix is to downgrade to 7.0.5 until the problem with the Diversion/From header is fixed. (hopefully Q4 maint release).

But I believe I got it working with this setup on 7.0.23:

SIP trunk setup:
Send Caller ID: Diversion
Local URI: *
Contact: *
Display Name *
PAI: None

Incoming Call Route:
Incoming Number: (DID for ext 665)
Destination: 665

Users (665):
Check Forward Unconditional
Forward Number: 718452225960
Uncheck Mobile Twinning (This workaround does not apply to Twinning)

If 665 does not have a DID number, you may need to change the 7 short code to include a valid 10 digit number like this:

Code: 7N;
Feature: Dial
Telephone Number: Ns8455551234
Line Group ID: (SIP trunk group)

 

[HarryNorstarGuy]

Thanks - I will try that with a DID.

All of my tests above involved dialing in to a Main # answered by AA on VM-Pro. Trx to 665.

 

[HarryNorstarGuy]

Hello RedPhone

I tried to set everything as you mentioned above, it works fine on DID, but not is we dial in to the main number and dial the Extension, which brings me to ask if the issue can't perhaps be resolved by adding something in VM PRO in the transfer node to pass along CID so that the system would treat a forwarded call the same way it does for direct DID calls.

again - any help would be greatly appreciated.

cheers

 

[AACon]

I wonder if putting $CLI in the source of the transfer would help at all.

-Austin
ACE: Implement IP Office

 

[HarryNorstarGuy]

I checked and the $CLI is in the source of the transfer.