[Asterisk-Users] SIP trunk: asterisk - callmanager

[Tim Connolly]

Yeah I know, its an old post, but I have just the OPPOSITE problem. I can
all out from my Cisco SIP phones across the SIP trunk (CCM -> *) but not the
reverse. Any help would be greatly appreciated...

Sip.conf ------------
[labcm33]
type=friend
host=1.2.3.4
context=incoming
disallow=all
allow=ulaw 
allow=alaw
nat=no
canreinvite=yes
qualify=yes

Extensions.conf --------------
[outgoing]
exten => _14XXX,1,ChanIsAvail(SIP/labcm33)
exten => _14XXX,2,Cut(AVAILCHAN=AVAILCHAN,,1)
exten => _14XXX,3,Dial(${AVAILCHAN},${ARG1})
exten => _14XXX,4,Hangup
exten => i,1,Congestion


I tried several variations in extensions.conf from an example taken from the
wiki: http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration
Ultimately, I guess I don't know the format of the URL CCM is expecting:
exten => _14XXX,3,Dial(${AVAILCHAN},${ARG1})  returns:


Sip read: 
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK290ae2a3;rport
From: "Tim" <sip:6101 at 5.6.7.8>;tag=as2fdd9958
To: <sip:1.2.3.4>;tag=16777270
Date: Fri, 04 Mar 2005 15:56:47 GMT
Call-ID: 26708fe6596e8f84219c185d442b414e at 5.6.7.8
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


9 headers, 0 lines
    -- Got SIP response 400 "Bad Request - 'Malformed/Missing URL'" back
from 1.2.3.4
Transmitting:> 
ACK sip:1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK290ae2a3;rport
From: "Tim Connolly" <sip:6101 at 5.6.7.8>;tag=as2fdd9958
To: <sip:1.2.3.4>;tag=16777270
Contact: <sip:6101 at 5.6.7.8>
Call-ID: 26708fe6596e8f84219c185d442b414e at 5.6.7.8
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 1.2.3.4:5062
    -- SIP/labcm33-69a9 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Hangup("SIP/6101-b03a", "") in new stack
  == Spawn extension (default, 14001, 4) exited non-zero on 'SIP/6101-b03a'
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
3.4.5.6:38887;branch=z9hG4bKac10dc3d00006f354228841c00001e070000ff7f
From: "Tim"<sip:6101 at myasteriskbox>;tag=10699268716550
To: <sip:14001 at myasteriskbox>;tag=as6b37c0b8
Call-ID: D214C82C-68FC-436F-A5FD-29E36ED29EEB at 172.16.220.61
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:14001 at 5.6.7.8>
Content-Length: 0


<...snip...?


403...circuit busy.. blah!

Any ideas?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David Kemp
Sent: Tuesday, October 19, 2004 9:22 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] SIP trunk: asterisk - callmanager

Pavel, have you resolved the CCM issue?

I have the same problem, I can place calls from Asterisk to CCM but not the
other way, same zero tcpdump when going CCM -> Asterisk?  Think it is
something to do with the CCM Media Termination Point but all shows OK.
Reams of CCM logs don't really say what is going on. Any ideas???

Dave


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at 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