[Asterisk-Users] change the caller id number

[Schweizer Laurent]

Hello,

 

I have this configuration

 

Cisco 2600 <-- --> SER <-- --> 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:0787518551 at x.x.x.170>;tag=4E95FB8E-264A..To:
<sip:4202372 at spadatel.org>..Date: Tue, 04 May 1993 23:16:59 GMT..Call-ID:
4A402DE8-47FC11CC-8701BFDF-F52BB803 at x.x.x.170..Supported:
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:0787518551 at x.x.x.170:5060>..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:0787518551 at x.x.x.170>;tag=4E95FB8E-264A..To:
<sip:4202372 at spadatel.org>;tag=as1ad1575c..Call-ID:
4A402DE8-47FC11CC-8701BFDF-F52BB803 at x.x.x.170..CSeq: 101 INVITE..User-Agent:
Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact:
<sip:0244202372 at x.x.x.215>..Content-Length: 0....

 

 

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

INVITE sip:a0244202372 at 10.192.72.197:5060 SIP/2.0..

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

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

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

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

Call-ID: 587559230e84dcee74a92a0562f90827 at x.x.x.215.

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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