[Asterisk-Users] Cisco Asterisk Integration

Dinesh dinesh at imcb.a-star.edu.sg
Thu Dec 2 19:57:38 MST 2004 

Additional SIP Debug.

What I don't understand is, why is the asterisk saying callmanager is busy?
When I have no channels between them?

Everyone is busy/congested at this time

Dinesh.

owl*CLI> 

Sip read: 
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.217.81.111:5060;branch=z9hG4bK097f501e
From: "2202" <sip:2202 at 10.217.81.111>;tag=as7e412e33
To: <sip:65869804 at 10.217.84.12>;tag=34032050
Date: Fri, 03 Dec 2004 02:54:07 GMT
Call-ID: 57c009ce0413ec6a5961285376a8774c at 10.217.81.111
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


9 headers, 0 lines
    -- Got SIP response 404 "Not Found" back from 10.217.84.12
Transmitting:
ACK sip:65869804 at 10.217.84.12 SIP/2.0
Via: SIP/2.0/UDP 10.217.81.111:5060;branch=z9hG4bK097f501e
From: "2202" <sip:2202 at 10.217.81.111>;tag=as7e412e33
To: <sip:65869804 at 10.217.84.12>;tag=34032050
Contact: <sip:2202 at 10.217.81.111>
Call-ID: 57c009ce0413ec6a5961285376a8774c at 10.217.81.111
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 10.217.84.12:5060
    -- SIP/callman02-f4bb is circuit-busy
  == Everyone is busy/congested at this time
    -- Executing Hangup("SIP/2202-1878", "") in new stack
  == Spawn extension (macro-dialout-callmanager, s, 4) exited non-zero on
'SIP/2202-1878' in macro 'dialout-callmanager'
  == Spawn extension (from-sip-internal, 65869804, 1) exited non-zero on
'SIP/2202-1878'
    -- Executing Hangup("SIP/2202-1878", "") in new stack
  == Spawn extension (from-sip-internal, h, 1) exited non-zero on
'SIP/2202-1878'
Reliably Transmitting (NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.217.64.92:5060;received=10.217.64.92;rport=5060
From: <sip:2202 at 10.217.81.111;user=phone>;tag=2319133461
To: <sip:65869804 at 10.217.81.111;user=phone>;tag=as6128b970
Call-ID: 687188633 at 10.217.64.92
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:65869804 at 10.217.81.111>
Content-Length: 0


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dinesh
Sent: Friday, December 03, 2004 10:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Cisco Asterisk Integration

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 at 10.217.81.111;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.217.64.92:5060
From: <sip:2202 at 10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804 at 10.217.81.111;user=phone>
Call-ID: 1878611129 at 10.217.64.92
CSeq: 1 INVITE
Contact: <sip:2202 at 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 at 10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804 at 10.217.81.111;user=phone>;tag=as6307c481
Call-ID: 1878611129 at 10.217.64.92
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9804 at 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 at 10.217.64.92' in 15000 ms
Found user '2202'
owl*CLI> 

Sip read: 
ACK sip:9804 at 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 at 10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804 at 10.217.81.111;user=phone>;tag=as6307c481
Call-ID: 1878611129 at 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 at 10.217.81.111;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.217.64.92:5060
From: <sip:2202 at 10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804 at 10.217.81.111;user=phone>
Call-ID: 1878611129 at 10.217.64.92
CSeq: 2 INVITE
Contact: <sip:2202 at 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 at 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 at 10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804 at 10.217.81.111;user=phone>;tag=as6307c481
Call-ID: 1878611129 at 10.217.64.92
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9804 at 10.217.81.111>
Content-Length: 0


 to 10.217.64.92:5060
owl*CLI> 

Sip read: 
ACK sip:9804 at 10.217.81.111;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.217.64.92:5060
From: <sip:2202 at 10.217.81.111;user=phone>;tag=4060751856
To: <sip:9804 at 10.217.81.111;user=phone>;tag=as6307c481
Call-ID: 1878611129 at 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 at 10.217.81.11
1",response="b9de8dd79e84050eff121e18970f3f58"
Content-Length: 0


9 headers, 0 lines
Destroying call '1878611129 at 10.217.64.92'




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