Paul Davidson
2004-Nov-16 22:00 UTC
[Asterisk-Users] Connection of Asterisk to Cisco Callmanager via H.323
Greetings- I've managed to successfully integrate our Cisco Callmanager (v3.3.3) to Asterisk for the purposes of replacing a broken and expensive meetme system with one that works. Under CCM v3.3.3, there is no support for SIP trunks- our 4.0 migration is still some time off. I'm using the GnuGK Open Gatekeeper to handle the signalling, and I've got an H.323 trunk set up. Calls from the Cisco side into Asterisk work swimmingly- very smooth. (kudos to Mark). I'm using the NuFone H.323 channel driver, and OH323 1.12.2. I'm running Asterisk 1.0 RC2. However, here's the rub. For some reason, I cannot get calls back from Asterisk to the Callmanager. If I use OhPhone pointed directly at the gatekeeper, I can get it to work- so I've ruled out Callmanager or Gatekeeper setups- but calls from Asterisk to Callmanager (using DIAX or other softphone authenticated to Asterisk) fail- the softphone keeps ringing, but the phone doesn't. DIAX will make successful calls into Asterisk for other numbers, so I've got the IAX2 authentication apparently working- but there's a problem somewhere (I believe) in the signalling from Asterisk to Callmanager. Note that I've got the prefix 781 set up to route to the Callmanager- feed 781XXXX to the gatekeeper via OhPhone, and extension XXXX rings on my desk. From tcpdump and ethereal, I'm seeing a Q.931 SETUP message going back to the Callmanager from Asterisk when I attempt the call- it seems to have the correct information, but differs in some key ways from the working OhPhone call- some IE's are missing (calling party, for instance- though called party is there and correct), and the Bearer Capability is set to 'Speech' out of Asterisk, and 'Unrestricted digital information' out of OhPhone. I've included relevant snippets of my h323.conf and extensions.conf below- has anyone sucessfully done this? Once I get it 100% working, I'll fill in the details back to the Wiki- there appears to be a working example based on SIP there already, with the ominous note 'Upgrade to 1.0 and it will work under H.323 too!'. Not very helpful, I'm afraid. Thanks in advance- snippets follow. -Paul Davidson ------ snip h323.conf sanitized for your protection - IP address replaced with XX------ [general] ; ; Port to listen to port=1720 bindaddr = XX.XX.XX.105 gatekeeper=XX.XX.XX.112 disallow=all allow=ulaw context=voip-h323 ; ; Specify alias(es) of this host. ; It may be used multiple times. ; alias=AST ;alias=4489 ; ; Set the context of H323 calls ; [conference] type=h323 e164=4488 context=voip-h323 ------ snip extensions.conf ---------- [default] ; include => voip-h323 ; Dial pattern to get back to Cisco Callmanager exten => _781XXXX,1,Dial(H323/${EXTEN}@192.168.222.1) exten => _781XXXX,2,Congestion