Hello,

 

I have this configuration

 

Cisco 2600 ß à SER ß à Asterisk

 

 

When I receive a call on asterisk from ser then I dial 2 different extensions a${EXTEN} and b${EXTEN}  but I can not set correctly the caller id number.

 

When I make a dial asterisk set caller id name and number to “asterisk”. I can change the name with SetCIDName but it is not working for the caller Id number, I have tried to use SetCIDNum.

 

Laurent

 

 

U x.x.x.213:5060 -> x.x.x.215:5060

 INVITE sip:0244202372@ x.x.x.215 SIP/2.0..Record-Route: <sip:0244202372@ x.x.x.213;ftag=4E95FB8E-264A;lr=on>..Record-Route: <sip:4202372@ x.x.x.213;ftag=4E95FB8E-264A;lr=on>..Via: SIP/2.0/

UDP x.x.x.213;branch=z9hG4bK2f25.b2aa926.0..Via: SIP/2.0/UDP x.x.x.214;branch=z9hG4bK2f25.3c1833e7.0..Via: SIP/2.0/UDP x.x.x.213;branch=z9hG4bK2f25.a2aa926.0..Via: SIP/2.0/UDP  x.x.x.170:5060..From: <sip:[EMAIL PROTECTED]>;tag=4E95FB8E-264A..To: <sip:[EMAIL PROTECTED]>..Date: Tue, 04 May 1993 23:16:59 GMT..Call-ID: [EMAIL PROTECTED]: timer,100rel..Min-SE:  1800..Cisco-Guid: 1245359493-1207701964-2264842207-4113283075..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBS

CRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 3..Timestamp: 736557419..Contact: <sip:[EMAIL PROTECTED]:5060>..Expires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..C

ontent-Length: 360..P-hint: usrloc applied....v=0..o=CiscoSystemsSIP-GW-UserAgent 3339 2725 IN IP4 x.x.x.170..s=SIP Call..c=IN IP4 x.x.x.170..t=0 0..m=audio 16916 RTP/AVP 18 4 0 8 101..c=IN I

P4 x.x.x.170..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:4 G723/8000..a=fmtp:4 annexa=yes..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..

 

 

U x.x.x.215:5060 -> x.x.x.213:5060

  SIP/2.0 100 Trying..Via: SIP/2.0/UDP x.x.x.213;branch=z9hG4bK2f25.b2aa926.0..Via: SIP/2.0/UDP x.x.x.214;branch=z9hG4bK2f25.3c1833e7.0..Via: SIP/2.0/UDP x.x.x.213;branch=z9hG4bK2f25.a2aa

  926.0..Via: SIP/2.0/UDP x.x.x.170:5060..From: <sip:[EMAIL PROTECTED]>;tag=4E95FB8E-264A..To: <sip:[EMAIL PROTECTED]>;tag=as1ad1575c..Call-ID: [EMAIL PROTECTED]: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:[EMAIL PROTECTED]>..Content-Length: 0....

 

 

U x.x.x.215:5060 -> x.x.x.213:5061

INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0..

Via: SIP/2.0/UDPx.x.x.215:5060;branch=z9hG4bK6d41ce89;rport..

From: "asterisk" <sip:asterisk@x.x.x.215>;tag=as68aa6c65..

To: <sip:a0244202372 @10.192.72.197:5060>..

Contact: <sip:asterisk@x.x.x.215>..

Call-ID: [EMAIL PROTECTED].

CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Thu, 17 Feb 2005 10:29:46 GMT..

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

Content-Type: application/sdp.

Content-Length: 369...

v=0..o=root 32675 32675 IN IP4 x.x.x.215..s=session..

c=IN IP4 x.x.x.215..

t=0 0..

m=audio  19062 RTP/AVP 0 8 4 18 3 101.

a=rtpmap:0 PCMU/8000.

 

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