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:[EMAIL PROTECTED]:5060 SIP/2.0 005 Via: SIP/2.0/UDP 10.217.84.12:5060;branch=z9hG4bK4709768 006 From: "Dinesh" <sip:[EMAIL PROTECTED]>;tag=34015864 007 To: <sip:[EMAIL PROTECTED]> 008 Date: Wed, 01 Dec 2004 03:37:53 GMT 009 Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;party=calling;screen=no;privacy=off 017 Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=34015864 051 To: <sip:[EMAIL PROTECTED]>;tag=as2fdffb5d 052 Call-ID: [EMAIL PROTECTED] 053 CSeq: 101 INVITE 054 User-Agent: Asterisk PBX 055 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 056 Contact: <sip:[EMAIL PROTECTED]> 057 Content-Length: 0 058 059 060 to 10.217.84.12:5060 061 owl*CLI> 062 063 Sip read: 064 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 065 Via: SIP/2.0/UDP 10.217.84.12:5060;branch=z9hG4bK4709768 066 From: "Dinesh" <sip:[EMAIL PROTECTED]>;tag=34015864 067 To: <sip:[EMAIL PROTECTED]>;tag=as2fdffb5d 068 Date: Wed, 01 Dec 2004 03:37:53 GMT 069 Call-ID: [EMAIL PROTECTED] 070 Max-Forwards: 6 071 CSeq: 101 ACK 072 Content-Length: 0 073 074 075 9 headers, 0 lines 076 Destroying call '[EMAIL PROTECTED]' 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 : [EMAIL PROTECTED] WWW: www.imcb.a-star.edu.sg _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
