Hello All, I have managed to get my cisco and asterisk able to talk to one another I think. But cannot make a call from a phone behind call manager to the asterisk server. I have followed the cisco asterisk integration on the wiki. I have also setup a number 3000 for dialing for current local time and date on asterisk. I can call from a sip phone behind asterisk, no problems. The problem occurs when I call from a phone behind cisco call manager. I have set up route pattern to divert all calls to the asterisk if the user presses 7.! . Anyone help would be appreciated:) This is the debug message I am getting when I dial 3000 from a cisco phone behind call manager. 001 owl*CLI> 002 003 Sip read: 004 INVITE sip:3000@10.217.81.111:5060 SIP/2.0 005 Via: SIP/2.0/UDP 10.217.84.12:5060;branch=z9hG4bK4709768 006 From: "Dinesh" <sip:65869804@10.217.84.12>;tag=34015864 007 To: <sip:3000@10.217.81.111> 008 Date: Wed, 01 Dec 2004 03:37:53 GMT 009 Call-ID: 607c8400-1da1614d-4262-c54d90a@10.217.84.12 010 Supported: timer 011 Min-SE: 360 012 User-Agent: Cisco-CCM4.0 013 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK 014 CSeq: 101 INVITE 015 Max-Forwards: 6 016 Remote-Party-ID: "Dinesh" <sip:65869804@10.217.84.12>;party=calling;screen=no;privacy=off 017 Contact: <sip:65869804@10.217.84.12:5060> 018 Expires: 180 019 Allow-Events: telephone-event 020 Content-Type: application/sdp 021 Content-Length: 227 022 023 v=0 024 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 10.217.84.12 025 s=SIP Call 026 c=IN IP4 10.217.84.11 027 t=0 0 028 m=audio 25182 RTP/AVP 0 101 029 a=sendrecv 030 a=rtpmap:0 PCMU/8000 031 a=ptime:20 032 a=rtpmap:101 telephone-event/8000 033 a=fmtp:101 0-15 034 035 18 headers, 11 lines 036 Using latest request as basis request 037 Sending to 10.217.84.12 : 5060 (non-NAT) 038 Found RTP audio format 0 039 Found RTP audio format 101 040 Peer audio RTP is at port 10.217.84.11:25182 041 Found description format PCMU 042 Found description format telephone-event 043 Capabilities: us - 0xc(ULAW|ALAW), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW) 044 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) 045 Found peer 'callman02' 046 Looking for 3000 in from-sip-external 047 Reliably Transmitting (no NAT): 048 SIP/2.0 404 Not Found 049 Via: SIP/2.0/UDP 10.217.84.12:5060;branch=z9hG4bK4709768 050 From: "Dinesh" <sip:65869804@10.217.84.12>;tag=34015864 051 To: <sip:3000@10.217.81.111>;tag=as2fdffb5d 052 Call-ID: 607c8400-1da1614d-4262-c54d90a@10.217.84.12 053 CSeq: 101 INVITE 054 User-Agent: Asterisk PBX 055 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 056 Contact: <sip:3000@10.217.81.111> 057 Content-Length: 0 058 059 060 to 10.217.84.12:5060 061 owl*CLI> 062 063 Sip read: 064 ACK sip:3000@10.217.81.111:5060 SIP/2.0 065 Via: SIP/2.0/UDP 10.217.84.12:5060;branch=z9hG4bK4709768 066 From: "Dinesh" <sip:65869804@10.217.84.12>;tag=34015864 067 To: <sip:3000@10.217.81.111>;tag=as2fdffb5d 068 Date: Wed, 01 Dec 2004 03:37:53 GMT 069 Call-ID: 607c8400-1da1614d-4262-c54d90a@10.217.84.12 070 Max-Forwards: 6 071 CSeq: 101 ACK 072 Content-Length: 0 073 074 075 9 headers, 0 lines 076 Destroying call '607c8400-1da1614d-4262-c54d90a@10.217.84.12' 077 owl*CLI> exit owl*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status 2202/2202 10.217.64.92 D N 255.255.255.255 5060 Unmonitored 2201/2201 (Unspecified) D N 255.255.255.255 0 UNKNOWN callman02 10.217.84.12 255.255.255.255 5060 OK (41 ms) callman01 10.217.84.11 255.255.255.255 5060 OK (41 ms) regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : dinesh@imcb.a-star.edu.sg WWW: www.imcb.a-star.edu.sg
On Wed, 2004-12-01 at 11:43 +0800, Dinesh wrote:> > 046 Looking for 3000 in from-sip-external > 047 Reliably Transmitting (no NAT): > 048 SIP/2.0 404 Not FoundLooks to me like extension 3000 is not in the "from-sip-external" context. Check your dialplan. Jeff -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041130/dd1a5ba1/attachment.pgp
Having managed to fix that, I have a new problem.
Calls are transferred to asterisk by pressing 7, as per the route plan on
cisco call manager. When I dial a extension on asterisk say for example
"2202" on a cisco phone behind call manager by pressing
"72202" the caller
id I get on the cisco phone behind asterisk is my full number phone number.
I can get the call, but when I try to return the call.
[macro-dialout-callmanager]
exten => s,1,ChanIsAvail(SIP/callman02&SIP/callman01)
exten => s,2,Cut(AVAILCHAN=AVAILCHAN,,1)
exten => s,3,Dial(${AVAILCHAN}/${ARG1})
exten => s,4,Hangup
exten => s,102,Congestion
[outgoing]
exten => _9804,1,Macro(dialout-callmanager,${EXTEN})
exten => _65869804,1,Macro(dialout-callmanager,${EXTEN})
I am trying to make asterisk dial my extension 9804 call manager.
When I do this error message below. I understand that
Found user '2202'
Looking for 9804 in from-sip-internal
Reliably Transmitting (NAT):
SIP/2.0 404 Not Found
But, my context for the call manager is also from-sip-internal. Can anyone
help me what I am doing wrong?
[callman01]
type=friend
context=from-sip-internal
regards,
Dinesh.
owl*CLI>
Sip read:
INVITE sip:9804@10.217.81.111;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.217.64.92:5060
From: <sip:2202@10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804@10.217.81.111;user=phone>
Call-ID: 1878611129@10.217.64.92
CSeq: 1 INVITE
Contact: <sip:2202@10.217.64.92:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7905/1.02-040406A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Expires: 300
Content-Length: 279
Content-Type: application/sdp
v=0
o=2202 30534 30534 IN IP4 10.217.64.92
s=Cisco 7905 SIP Call
c=IN IP4 10.217.64.92
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
12 headers, 12 lines
Using latest request as basis request
Sending to 10.217.64.92 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.217.64.92:16384
Found description format PCMU
Found description format G729
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc(ULAW|ALAW), peer -
audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.217.64.92:5060;received=10.217.64.92;rport=5060
From: <sip:2202@10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804@10.217.81.111;user=phone>;tag=as6307c481
Call-ID: 1878611129@10.217.64.92
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9804@10.217.81.111>
Proxy-Authenticate: Digest realm="asterisk",
nonce="2a695c7a"
Content-Length: 0
to 10.217.64.92:5060
Scheduling destruction of call '1878611129@10.217.64.92' in 15000 ms
Found user '2202'
owl*CLI>
Sip read:
ACK sip:9804@10.217.81.111;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.217.64.92:5060;received=10.217.64.92;rport=5060
From: <sip:2202@10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804@10.217.81.111;user=phone>;tag=as6307c481
Call-ID: 1878611129@10.217.64.92
CSeq: 1 ACK
User-Agent: Cisco-CP7905/1.02-040406A
Content-Length: 0
8 headers, 0 lines
owl*CLI>
Sip read:
INVITE sip:9804@10.217.81.111;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.217.64.92:5060
From: <sip:2202@10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804@10.217.81.111;user=phone>
Call-ID: 1878611129@10.217.64.92
CSeq: 2 INVITE
Contact: <sip:2202@10.217.64.92:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7905/1.02-040406A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Proxy-Authorization: Digest
username="2202",realm="asterisk",nonce="2a695c7a",uri="sip:9804@10.217.81.11
1",response="b9de8dd79e84050eff121e18970f3f58"
Expires: 300
Content-Length: 279
Content-Type: application/sdp
v=0
o=2202 30554 30554 IN IP4 10.217.64.92
s=Cisco 7905 SIP Call
c=IN IP4 10.217.64.92
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 12 lines
Using latest request as basis request
Sending to 10.217.64.92 : 5060 (NAT)
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.217.64.92:16384
Found description format PCMU
Found description format G729
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc(ULAW|ALAW), peer -
audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Found user '2202'
Looking for 9804 in from-sip-internal
Reliably Transmitting (NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.217.64.92:5060;received=10.217.64.92;rport=5060
From: <sip:2202@10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804@10.217.81.111;user=phone>;tag=as6307c481
Call-ID: 1878611129@10.217.64.92
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9804@10.217.81.111>
Content-Length: 0
to 10.217.64.92:5060
owl*CLI>
Sip read:
ACK sip:9804@10.217.81.111;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.217.64.92:5060
From: <sip:2202@10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804@10.217.81.111;user=phone>;tag=as6307c481
Call-ID: 1878611129@10.217.64.92
CSeq: 2 ACK
User-Agent: Cisco-CP7905/1.02-040406A
Proxy-Authorization: Digest
username="2202",realm="asterisk",nonce="2a695c7a",uri="sip:9804@10.217.81.11
1",response="b9de8dd79e84050eff121e18970f3f58"
Content-Length: 0
9 headers, 0 lines
Destroying call '1878611129@10.217.64.92'
Hello all, Just to let you all know I fixed it with some help from Jeffrey and Damian. It was due to the calling search space on call manager. When I sorted that out, it was working fine. Dinesh. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dinesh Sent: Wednesday, December 01, 2004 11:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco Asterisk Integration Hello All, I have managed to get my cisco and asterisk able to talk to one another I think. But cannot make a call from a phone behind call manager to the asterisk server. I have followed the cisco asterisk integration on the wiki. I have also setup a number 3000 for dialing for current local time and date on asterisk. I can call from a sip phone behind asterisk, no problems. The problem occurs when I call from a phone behind cisco call manager. I have set up route pattern to divert all calls to the asterisk if the user presses 7.! . Anyone help would be appreciated:) This is the debug message I am getting when I dial 3000 from a cisco phone behind call manager. 001 owl*CLI> 002 003 Sip read: 004 INVITE sip:3000@10.217.81.111:5060 SIP/2.0 005 Via: SIP/2.0/UDP 10.217.84.12:5060;branch=z9hG4bK4709768 006 From: "Dinesh" <sip:65869804@10.217.84.12>;tag=34015864 007 To: <sip:3000@10.217.81.111> 008 Date: Wed, 01 Dec 2004 03:37:53 GMT 009 Call-ID: 607c8400-1da1614d-4262-c54d90a@10.217.84.12 010 Supported: timer 011 Min-SE: 360 012 User-Agent: Cisco-CCM4.0 013 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK 014 CSeq: 101 INVITE 015 Max-Forwards: 6 016 Remote-Party-ID: "Dinesh" <sip:65869804@10.217.84.12>;party=calling;screen=no;privacy=off 017 Contact: <sip:65869804@10.217.84.12:5060> 018 Expires: 180 019 Allow-Events: telephone-event 020 Content-Type: application/sdp 021 Content-Length: 227 022 023 v=0 024 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 10.217.84.12 025 s=SIP Call 026 c=IN IP4 10.217.84.11 027 t=0 0 028 m=audio 25182 RTP/AVP 0 101 029 a=sendrecv 030 a=rtpmap:0 PCMU/8000 031 a=ptime:20 032 a=rtpmap:101 telephone-event/8000 033 a=fmtp:101 0-15 034 035 18 headers, 11 lines 036 Using latest request as basis request 037 Sending to 10.217.84.12 : 5060 (non-NAT) 038 Found RTP audio format 0 039 Found RTP audio format 101 040 Peer audio RTP is at port 10.217.84.11:25182 041 Found description format PCMU 042 Found description format telephone-event 043 Capabilities: us - 0xc(ULAW|ALAW), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW) 044 Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) 045 Found peer 'callman02' 046 Looking for 3000 in from-sip-external 047 Reliably Transmitting (no NAT): 048 SIP/2.0 404 Not Found 049 Via: SIP/2.0/UDP 10.217.84.12:5060;branch=z9hG4bK4709768 050 From: "Dinesh" <sip:65869804@10.217.84.12>;tag=34015864 051 To: <sip:3000@10.217.81.111>;tag=as2fdffb5d 052 Call-ID: 607c8400-1da1614d-4262-c54d90a@10.217.84.12 053 CSeq: 101 INVITE 054 User-Agent: Asterisk PBX 055 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 056 Contact: <sip:3000@10.217.81.111> 057 Content-Length: 0 058 059 060 to 10.217.84.12:5060 061 owl*CLI> 062 063 Sip read: 064 ACK sip:3000@10.217.81.111:5060 SIP/2.0 065 Via: SIP/2.0/UDP 10.217.84.12:5060;branch=z9hG4bK4709768 066 From: "Dinesh" <sip:65869804@10.217.84.12>;tag=34015864 067 To: <sip:3000@10.217.81.111>;tag=as2fdffb5d 068 Date: Wed, 01 Dec 2004 03:37:53 GMT 069 Call-ID: 607c8400-1da1614d-4262-c54d90a@10.217.84.12 070 Max-Forwards: 6 071 CSeq: 101 ACK 072 Content-Length: 0 073 074 075 9 headers, 0 lines 076 Destroying call '607c8400-1da1614d-4262-c54d90a@10.217.84.12' 077 owl*CLI> exit owl*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status 2202/2202 10.217.64.92 D N 255.255.255.255 5060 Unmonitored 2201/2201 (Unspecified) D N 255.255.255.255 0 UNKNOWN callman02 10.217.84.12 255.255.255.255 5060 OK (41 ms) callman01 10.217.84.11 255.255.255.255 5060 OK (41 ms) regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : dinesh@imcb.a-star.edu.sg WWW: www.imcb.a-star.edu.sg _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users