Solved: How to pass ISDN Caller ID Name to 3640 sip-ua

penn · ‎11-13-2006

I am not able to get Caller ID name to pass from a ISDN PRI connected to a 3640 (12.4 ios) through to the sip-ua. Caller ID number works & passes fine. I see the caller ID name come through on isdnq931 and ccsip debugs as shown below:

Nov 13 17:56:29 CDT: ISDN Se1/1:23 Q931: RX <- SETUP pd = 8 callref = 0x5C19

Bearer Capability i = 0x9090A2

Standard = CCITT

Transfer Capability = 3.1kHz Audio

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA18381

Preferred, Channel 1

Facility i = 0x9F8B0100A115020101020100800D505249534D4E45542020494E43

Protocol Profile = Networking Extensions

0xA115020101020100800D505249534D4E45542020494E43

Component = Invoke component

Invoke Id = 1

Operation = CallingName

Name presentation allowed

Name = PRISMNET INC

Progress Ind i = 0x8283 - Origination address is non-ISDN

Calling Party Number i = 0x2183, '5128219133'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '5126470901'

Plan:ISDN, Type:National

Nov 13 17:56:29 CDT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:5126470901@X.X.X.X:5060 SIP/2.0

Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK646B1BF1

From: <sip:5128219133@Y.Y.Y.Y>;tag=6FBE25B4-6A5

To: <sip:5126470901@X.X.X.X>

Date: Mon, 13 Nov 2006 22:56:29 GMT

Call-ID: 8A82829-72A111DB-934AA54B-2A6A3B55@Y.Y.Y.Y

Supported: 100rel,timer,replaces

Min-SE: 1800

Cisco-Guid: 144998049-1923158491-2832072705-1108514560

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY

, INFO, UPDATE, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Remote-Party-ID: <sip:5128219133@Y.Y.Y.Y>;party=calling;screen=yes;privacy=off

Timestamp: 1163458589

Contact: <sip:5128219133@Y.Y.Y.Y:5060>

Expires: 300

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 253

From what I can only guess, the

<sip:5128219133@Y.Y.Y.Y>;party=calling;screen=yes

(screen=yes) is the culprit. Does anyone know how to set the call screen=no, or how to make it pass the Caller Name through? Here's all I have for my sip-ua:

sip-ua

retry invite 3

retry response 3

retry bye 3

retry cancel 3

timers expires 300000

sip-server ipv4:X.X.X.X:5060

gogasca · ‎11-13-2006

The screening indicator is generally not used to decide whether to show the calling party's number|name at the called entity. Rather, it's purpose is to tell the called entity whether the calling party number was set by the calling user (and possibly verified by the network), or if the number was set by the network (for example if an invalid calling number was sent by the calling user and the network over-wrote it).

According to Q.951 clause 3, the originating local exchange is to discard any screening indicator it might receive from the calling user. This is of course not to be confused with the Presentation Indicator (Allowed/Restricted).

There is no way currently in IOS CLI to set the screening indication bit as far as I know. But these values can be set to customer-required values by using TCL script. Use the default "session" script in IOS gateway, modify it according to your requirement and then load the session application in the incoming dial peer of the gateway.

Use "set callinfo" to set the values of Octet 3a and then associate with the setup message using "leg setup" command.

IE:

set callInfo (originationNumSI) "usr_provided_screening_passed"

Refer the following links,

http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2/chapter3.htm#1036917

http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2/index.htm

"0" (user-provided, not screened) on "1" (user-provided, verified and passed).

SIP can support calling name in Facility-IE

Per your debug:

Facility i = 0x9F8B0100A115020101020100800D505249534D4E45542020494E43

The name is coming in a FACILITY message after the initial set up message so the INVITE needs to be buffered until the receipt of the calling name, we need to configure:

- "isdn supp-service name calling" under isdn d-channel interface

- need to configure under sip-ua

no remote-party-id

timers buffer-invite 5000 (or anything more than the timeperiod in which the FACILITY

comes in after the q931 SETUP).

Calling name should work now as a buffered INVITE is sent with the calling name in the

FROM field.

Let me know

//G

View solution in original post

gogasca · ‎11-13-2006

The screening indicator is generally not used to decide whether to show the calling party's number|name at the called entity. Rather, it's purpose is to tell the called entity whether the calling party number was set by the calling user (and possibly verified by the network), or if the number was set by the network (for example if an invalid calling number was sent by the calling user and the network over-wrote it).

According to Q.951 clause 3, the originating local exchange is to discard any screening indicator it might receive from the calling user. This is of course not to be confused with the Presentation Indicator (Allowed/Restricted).

There is no way currently in IOS CLI to set the screening indication bit as far as I know. But these values can be set to customer-required values by using TCL script. Use the default "session" script in IOS gateway, modify it according to your requirement and then load the session application in the incoming dial peer of the gateway.

Use "set callinfo" to set the values of Octet 3a and then associate with the setup message using "leg setup" command.

IE:

set callInfo (originationNumSI) "usr_provided_screening_passed"

Refer the following links,

http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2/chapter3.htm#1036917

http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/vapp_dev/tclivrv2/index.htm

"0" (user-provided, not screened) on "1" (user-provided, verified and passed).

SIP can support calling name in Facility-IE

Per your debug:

Facility i = 0x9F8B0100A115020101020100800D505249534D4E45542020494E43

The name is coming in a FACILITY message after the initial set up message so the INVITE needs to be buffered until the receipt of the calling name, we need to configure:

- "isdn supp-service name calling" under isdn d-channel interface

- need to configure under sip-ua

no remote-party-id

timers buffer-invite 5000 (or anything more than the timeperiod in which the FACILITY

comes in after the q931 SETUP).

Calling name should work now as a buffered INVITE is sent with the calling name in the

FROM field.

Let me know

//G

penn · ‎11-14-2006

That worked. You totally rock!!

gogasca · ‎11-13-2006

You may also want to check this good link:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftsipext.htm#wp1056966

Under:

sip-ua

calling-info pstn-to-sip