02-17-2005 02:28 PM - edited 03-13-2019 08:04 AM
Is there any design problem with running a few voip h.323 trunks from ccme to an h.323 gateway? The gateway would be a 1760 with some fxo ports to the pstn.
02-18-2005 12:18 PM
I shouldn't thinks so - just a matter of properly configuring your dial-peers etc...
02-21-2005 12:14 PM
Inbound/outbound calls work great, however, when I've got a call terminated and the voice path is pstn->1760gateway->2610ccme->ipphone when I transfer call to voicemail, it rings busy. If a call comes in on the second gateway and rolls to vm, it rings busy. Any sip dialpeer stuff that I should have?
02-21-2005 12:37 PM
Hi
This could be because VM requires G711, whereas G729 is default codec for voip dial-peers...
Try changing the dial-peers to :
codec g711ulaw
If this works, you have a couple of options:
1) If you have DSPs on your CME router spare, and have CME 3.2, I believe you can transcode for CUE sessions..
2) Set up a voice class codec with preference listed codecs, then apply it to your dial-peers:
Router#conf term
Router(config)#voice class codec 99
Router(config-class)#codec preference 1 g711ulaw
Router(config-class)#codec preference 2 g729br8
Router(config-class)#codec preference 3 g729r8
Router(config-class)#end
Router(config)#dial-peer voice 2000 voip
Router(config-dial-peer)#voice-class codec
Hope this helps...
Aaron
Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: