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New Member

Calling number, Sip Trunks and Asterisk .

Hi,

A 7960 forwards calls not answered to a number thats routed over a sip trunk to an Asterisk server. Asterisk sees the Caller Number as the 7960.

So when asterisk has played its various messages and forwards the call back to an AC broadcast queue i see the caller number as the original forwarding phone not the original calling number of the customer.

I'm guessing its an Asterisk issue but could it be SIP ? If i forward the phone to another CCM extension I do see the original calling number.

Matt.

1 ACCEPTED SOLUTION

Accepted Solutions
Green

Re: Calling number, Sip Trunks and Asterisk .

Im running CCM 4.2SR1

7960 CFNA --> SIP TRUNK --> * ---> extensions.conf --> Go to CCM again ---> we see 7960 not very first-

In the CCM Side in the SIP trunk make sure you have the following:

Outbound Calls

Calling Party Selection* Originator

Here in mylab works fine:

105 IPC --> 102 CFA to 210 --> in * 210 send it back to 101 and in 101 phone I do see 105.

***************************************************

Outgoing call leg

INVITE sip:210@110.10.200.2:5060 SIP/2.0

Via: SIP/2.0/UDP 110.10.200.3:5060;branch=z9hG4bK3fdd7fba

From: "IP Communicator" <105>;tag=16777280

To: <210>

Date: Tue, 09 May 2006 05:39:45 GMT

Call-ID: 373c2c00-1db1fd21-20-3c80a6e@110.10.200.3

Supported: timer

Min-SE: 1800

User-Agent: Cisco-CCM4.1

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK

CSeq: 101 INVITE

Max-Forwards: 70

Remote-Party-ID: "IP Communicator" <105>;party=calling;screen=no;privacy=off

Contact: <105>

Diversion: <102>;reason=unconditional

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 227

***************************************************

Incoming call leg

05/09/2006 00:39:45.500 CCM|Incoming UDP SIP message size 698 from 110.10.200.2:[5060]:

INVITE sip:101@110.10.200.3 SIP/2.0

Via: SIP/2.0/UDP 110.10.200.2:5060;branch=z9hG4bK0fdf17b2

From: "IP Communicator" <105>;tag=as3b97aa13

To: <101>

Contact: <105>

Call-ID: 5c721b3a211c47d57688f64440b23821@110.10.200.2

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Date: Tue, 09 May 2006 05:35:38 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Content-Type: application/sdp

Content-Length: 238

It works fine also for my PP (109). I used First Available algorithm and I do see Original calling party number. Can you give it a try?

;

exten => 210,1,Dial(SIP/109@CallManager)

Under CCMAdmin| Service | Service Parameters | TCD (Telephony Call Dispatcher)

Make sure you have these:

Redirect With Directing Party CSS* True

Reset Original Called Party on Redirect* True

HTH

//G

2 REPLIES
Green

Re: Calling number, Sip Trunks and Asterisk .

Im running CCM 4.2SR1

7960 CFNA --> SIP TRUNK --> * ---> extensions.conf --> Go to CCM again ---> we see 7960 not very first-

In the CCM Side in the SIP trunk make sure you have the following:

Outbound Calls

Calling Party Selection* Originator

Here in mylab works fine:

105 IPC --> 102 CFA to 210 --> in * 210 send it back to 101 and in 101 phone I do see 105.

***************************************************

Outgoing call leg

INVITE sip:210@110.10.200.2:5060 SIP/2.0

Via: SIP/2.0/UDP 110.10.200.3:5060;branch=z9hG4bK3fdd7fba

From: "IP Communicator" <105>;tag=16777280

To: <210>

Date: Tue, 09 May 2006 05:39:45 GMT

Call-ID: 373c2c00-1db1fd21-20-3c80a6e@110.10.200.3

Supported: timer

Min-SE: 1800

User-Agent: Cisco-CCM4.1

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK

CSeq: 101 INVITE

Max-Forwards: 70

Remote-Party-ID: "IP Communicator" <105>;party=calling;screen=no;privacy=off

Contact: <105>

Diversion: <102>;reason=unconditional

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 227

***************************************************

Incoming call leg

05/09/2006 00:39:45.500 CCM|Incoming UDP SIP message size 698 from 110.10.200.2:[5060]:

INVITE sip:101@110.10.200.3 SIP/2.0

Via: SIP/2.0/UDP 110.10.200.2:5060;branch=z9hG4bK0fdf17b2

From: "IP Communicator" <105>;tag=as3b97aa13

To: <101>

Contact: <105>

Call-ID: 5c721b3a211c47d57688f64440b23821@110.10.200.2

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Date: Tue, 09 May 2006 05:35:38 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Content-Type: application/sdp

Content-Length: 238

It works fine also for my PP (109). I used First Available algorithm and I do see Original calling party number. Can you give it a try?

;

exten => 210,1,Dial(SIP/109@CallManager)

Under CCMAdmin| Service | Service Parameters | TCD (Telephony Call Dispatcher)

Make sure you have these:

Redirect With Directing Party CSS* True

Reset Original Called Party on Redirect* True

HTH

//G

New Member

Re: Calling number, Sip Trunks and Asterisk .

Thanks that works perfectly :)

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