Hi Guys,
I have some problems using Asterisk with my Cisco Callmanager. I´ve configured a Asterisk Server as an IVR System and at this point everything is working fine.
I can transfer calls from Callmanager via SIP-Trunk to the Asterisk Server. But when I want to route the call back to the Cisco Callmanager it won´t work. I get the following message when I debug the SIP Connection.
[code]tack
Audio is at 192.168.5.74 port 15964
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.5.61:5060:
INVITE sip:491534@DESOWI0020 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.74:5060;branch=z9hG4bK26c8987c;rport
From: “Pohle, Christian” sip:491536@192.168.5.74;tag=as70382076
To: sip:491534@DESOWI0020
Contact: sip:491536@192.168.5.74
Call-ID: 437819631bc845b44452ce26263ffc96@192.168.5.74
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 07 Nov 2008 10:08:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 2047 2047 IN IP4 192.168.5.74
s=session
c=IN IP4 192.168.5.74
t=0 0
m=audio 15964 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-- Called 491534@DESOWI0020
asterisk1*CLI>
<— SIP read from 192.168.5.61:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.74:5060;branch=z9hG4bK26c8987c;rport
From: “Pohle, Christian” sip:491536@192.168.5.74;tag=as70382076
To: sip:491534@DESOWI0020;tag=50871012
Date: Fri, 07 Nov 2008 10:10:18 GMT
Call-ID: 437819631bc845b44452ce26263ffc96@192.168.5.74
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
— (9 headers 0 lines) —
asterisk1*CLI>
<— SIP read from 192.168.5.61:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.5.74:5060;branch=z9hG4bK26c8987c;rport
From: “Pohle, Christian” sip:491536@192.168.5.74;tag=as70382076
To: sip:491534@DESOWI0020;tag=50871012
Date: Fri, 07 Nov 2008 10:10:18 GMT
Call-ID: 437819631bc845b44452ce26263ffc96@192.168.5.74
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Transmitting (no NAT) to 192.168.5.61:5060:
ACK sip:491534@DESOWI0020 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.74:5060;branch=z9hG4bK26c8987c;rport
From: “Pohle, Christian” sip:491536@192.168.5.74;tag=as70382076
To: sip:491534@DESOWI0020;tag=50871012
Contact: sip:491536@192.168.5.74
Call-ID: 437819631bc845b44452ce26263ffc96@192.168.5.74
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
-- SIP/DESOWI0020-08223060 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘SIP/192.168.5.62-0821f0e8’ status is 'CONGESTION’
Really destroying SIP dialog ‘437819631bc845b44452ce26263ffc96@192.168.5.74’ Method: INVITE
asterisk1*CLI>
<— SIP read from 192.168.5.62:5060 —>
BYE sip:891000@192.168.5.74:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.62:5060;branch=z9hG4bK31486290
From: “Pohle, Christian” sip:491536@192.168.5.62;tag=34433101
To: sip:891000@192.168.5.74;tag=as2f4c11e3
Date: Fri, 07 Nov 2008 10:10:08 GMT
Call-ID: 400b4580-1de1cab7-701c3-3e05a8c0@192.168.5.62
User-Agent: Cisco-CCM4.1
Max-Forwards: 70
CSeq: 102 BYE
Content-Length: 0[/code]
I used this HowTo to setup asterisk, so where is the Problem?
voip-info.org/wiki-Asterisk+ … ntegration
Is someone here who can help me out? PLEASE, I need some help!!!
thanks and regards
Christian