07-20-2006 04:05 AM - edited 03-13-2019 02:16 PM
Hello,
I have two cme's with unity expresses, when I make a call across the wan the call is fine however when the call goes to voice mail I see that it is forwarded but I can not hear the audio from unity.
07-20-2006 08:40 AM
Had a similar issue with the same setup. Ultimately fixed this by transfer-pattern commands. They were required for CME to hand off the call. 6100-6105 were various attendants and VM pilot numbers at the remote site. Remote extensions were in the 1400s with remote paging in the 400s.
Under telephony-service
transfer-pattern 1...
transfer-pattern 4..
transfer-pattern 61..
Your CME knows about it's own extensions but needs to be explicitly told which extensions it is allowed to transfer to. Until it's allowed to do the transfer, the remote CUE won't be contacted.
07-20-2006 09:09 AM
If the above doesn't fix your problem, try hard-coding G.711 as the codec under your dial-peer. By default they will use G.729 and I believe CUE can only do G.711.
07-20-2006 09:51 AM
The default-dial peer 0 will do any codec. Just make sure you have your remote dial-peer using G711. If you replaced your default-dial peer with a specific inbound dial-peer then the default is G729. Again make sure this is using g711.
Please rate any helpful posts
Thanks
Fred
07-20-2006 10:01 AM
I have tried the transfer-patterns although I did have a transfer-pattern .T already in the telephony-service. I am already forcing g711ulaw as the codec and have confirm this by way or call statistics. I can see that the call is transfered to the unity pilot number on the remote site but still no audio. If I call a number on the remote site that has no voicemail I am transfered to the operator, so I know unity express is pickup the call, but I still cannot get any audio out of the unity to a remote location.
Thanks
p-herr
07-20-2006 10:04 AM
If you are using H323 make sure you have voice services setup like below.
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
Please rate any helpful posts
Thanks
Fred
07-20-2006 10:07 AM
I already have the vioce service entries, as well as an h323-gateway bind on the lan ip.
p-herr
08-14-2006 07:19 AM
Hi i have the same problem but i am using an 2811 with MGCP configuration ... do you have any advice??? Thanks in advanced.
Pepe
08-15-2006 09:27 AM
I did solve this problem. It seems that
!
voice services voip
h323
call start slow
!
!
was the problem in my case. Removing the call start slow, fixed it.Hope that helps
ph
08-16-2006 10:36 AM
I tried to do this but the problem continue, anyboby can help me? i am using an mgcp configuration on the gateways ... and from the PSTN the callers van not hear the autoattendant express.
Thanks
08-17-2006 10:29 PM
I had the same problem, check the calling search spaces from the gateway ... maybe it hasn?t have the correct partitions.
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