[Asterisk-Users] Cisco Asterisk Integration

[Dinesh]

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 at 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 at 10.217.84.12>;tag=34015864
007 To: <sip:3000 at 10.217.81.111>
008 Date: Wed, 01 Dec 2004 03:37:53 GMT
009 Call-ID: 607c8400-1da1614d-4262-c54d90a at 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 at 10.217.84.12>;party=calling;screen=no;privacy=off
017 Contact: <sip:65869804 at 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 at 10.217.84.12>;tag=34015864
051 To: <sip:3000 at 10.217.81.111>;tag=as2fdffb5d
052 Call-ID: 607c8400-1da1614d-4262-c54d90a at 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 at 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 at 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 at 10.217.84.12>;tag=34015864
067 To: <sip:3000 at 10.217.81.111>;tag=as2fdffb5d
068 Date: Wed, 01 Dec 2004 03:37:53 GMT
069 Call-ID: 607c8400-1da1614d-4262-c54d90a at 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 at 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 at imcb.a-star.edu.sg
WWW: www.imcb.a-star.edu.sg