02-09-2007 04:53 AM - edited 03-14-2019 07:57 PM
Hi, the current situation is that we have 2 sites that each have 2811 routers and incoming PSTN lines. The sites are connected via a T1. Site 1 has an AIM-CUE that the calls from the PSTN at Site 1 and Site 2 are directed to for AA. In this scenaria assume all phones are registered at to the CME on the 2811 router at Site1. A call coming in from the PSTN at Site 2 will go to the AA, come out of the AA and be transfered to the correct extension but will not forward to VM on busy or no answer. When trying to hit VM the call gets droped. I would think that what I'm trying to accomplish should be feasible and I'm just missing something. So any and all help will be greatly appreciated.
Thank you,
Rob
02-09-2007 06:02 AM
Hi Rob,
This is just a guess, but it sure sounds like a codec/transcoding issue here. CUE only does G711 and the calls are probably coming in G729a.
Codecs and Transcoding
Cisco Unity Express supports only G.711 voice streams, so all calls made into the system (including those from IP phones, PSTN gateway ports and any other VoIP equipment in your network) must use G.711 if they enter the AA or voice mail pilot numbers that terminate on Cisco Unity Express.
If you require that G.729A calls traverse IP segments of your network between sites, and that these calls forward or dial direct into Cisco Unity Express AA or voice mail, then you must use a transcoding resource collocated with Cisco Unity Express to change the voice stream from G.729A to G.711 before the voice stream enters the AA or ***voice mail pilot numbers.
Hope this helps!
Rob
02-10-2007 09:50 AM
After a brief conversation with a very knowledgable fellow at TAC the solution became clear. What my understanding lacked was the proper way to implement h323. The solution is to use h323 with no supplementary h450.2 or h450.3 service. The dial-peer at Site 2 will then point to the CME at Site 1 and Site 1 perform hairpin routing.
Rob
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