I've an Auto Attendant number configured on a MCU MSE 8510 for conferencing.
The architecture is
H323 GW 15.3.3M
IP PHONE----CUCM 8.6.2
I've other stuff around the VCS-C but they're not involved here.
I've created on the MCU a conference number 7006500 that is registered as H323 and SIP onto the VCS-C (Gatekeeper and SIP Registrar)
I've set up a route pattern on my CUCM that points to the SIP Trunk to VCS-C for that number.
For internal calls it works perfectly. We can have video on IP Phones but I've set the region settings to no video for the moment.
But when a call comes from PSTN it's disconnected after 10s. I can hear the conference prompt asking for conference number and PIN, if I'm quick enough I can enter the conference also, so DTMF works. Then hang up.
Something related to codec for sure. Disconnect cause 47 in debug isdn q931.
When I check "Media Termination Point Required" on the SIP Trunk then it works for calls from PSTN. But - there is a but - if I let this "Media Termination Point Required" checked, then internal calls to MCU (that were working before) are almost unable to use the DTMF to enter conf code and Pin. MCU does not understand or detects with the utmost difficulties the keys pressed.
On MCU all the possible codecs are checked.
We advertise G722 internally and incoming calls from PSTN have a voice class codec applied for the dial-peers with G711 A and mu.
Here are the debugs
debug h225 asn1
debug h245 asn1
debug voip ccapi inout
Any idea would be appreciated.
I have looked at the edited trace file you sent and this has confirmed that you are doing fast start on H323 (even though I didnt see the OLC before the h225 setup. I assume its because you have edited the trace file) to Delayed offer on SIP. You need to either remove the inbound fast start on CUCM and configure slow start on your gateway or enable Early offer on the SIP trunk to the VCS (NB you will need an MTP for this)
To answer your question....
The default on H323 gateway is fast start. To enable fslow start on the gateway you will need to configure it under voice service voip or at the dial-peer level...
voice service voip
call start slow
voice class h323 1
h225 timeout tcp establish 3
call start slow
You then apply the voice class h323 command to the dial-peer pointing to cucm.
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"