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@x.x.x.170>;tag=4E95FB8E-264A..To:
<sip:4202372@spadatel.org>..Date: Tue, 04 May 1993 23:16:59 GMT..Call-ID:
4A402DE8-47FC11CC-8701BFDF-F52BB803@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@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@x.x.x.170>;tag=4E95FB8E-264A..To:
<sip:4202372@spadatel.org>;tag=as1ad1575c..Call-ID:
4A402DE8-47FC11CC-8701BFDF-F52BB803@x.x.x.170..CSeq: 101 INVITE..User-Agent:
Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact:
<sip:0244202372@x.x.x.215>..Content-Length: 0....
U x.x.x.215:5060 -> x.x.x.213:5061
INVITE sip:a0244202372@10.192.72.197:5060 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: 587559230e84dcee74a92a0562f90827@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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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@x.x.x.170>;tag=4E95FB8E-264A..To:
<sip:4202372@spadatel.org>..Date: Tue, 04 May 1993 23:16:59 GMT..Call-ID:
4A402DE8-47FC11CC-8701BFDF-F52BB803@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@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@x.x.x.170>;tag=4E95FB8E-264A..To:
<sip:4202372@spadatel.org>;tag=as1ad1575c..Call-ID:
4A402DE8-47FC11CC-8701BFDF-F52BB803@x.x.x.170..CSeq: 101 INVITE..User-Agent:
Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact:
<sip:0244202372@x.x.x.215>..Content-Length: 0....
U x.x.x.215:5060 -> x.x.x.213:5061
INVITE sip:a0244202372@10.192.72.197:5060 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: 587559230e84dcee74a92a0562f90827@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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