10-23-2013 05:59 AM - edited 03-16-2019 08:02 PM
I setup a call handler and during the greentings the caller should be able to enter anytime the extension. Everything looks fine, they can enter the extension during the greetings, the phone rings, the user picks it up, however there is no audio they cannot hear each other.
Using CUCM/CUC 8.2 with VG CUBE to the ITSP via SIP.
Any idea what it might be?
Thanks!
10-23-2013 09:54 AM
Does this work calling internally?
Do you see the phone receiving RTP in the statistics??
Have you checked for routing issues?
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10-24-2013 04:51 AM
could be codec negotiation as well (lack of transcoder), check you regions in cucm.
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10-24-2013 05:51 AM
Thanks for getting back...
Yes, it works internally.
I did not find this info using IPC, and I'm not able to access via web the phone in order to check the receiving RTP in the statistics. What's the right way to check this info? Is this command will help: sh voip rtp conn?
I'm going to double check for routing issue. I think I should issue: voice rtp send-recv command
I thought would be a codec negotiation too, but everything looks fine. The only thing I'm not sure if I would need the MTP checked. Now is not checked on the trunk in the CUCM.
Anyway, my guess would be routing issue and I'm going to apply this command, if it doesn't solve, then might be the codec issue. I'll let you know the results.
Once again thanks for your suggestions.
10-24-2013 08:27 AM
Codec issues usually drop the call after the signaling is done, meaning, it rings, but then you answer, the call is dropped, I don't think this is the case.
On the phones you can press the ? button twice to get statistics from current call.
HTH
java
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10-24-2013 02:04 PM
The only way I was able to get this solved was checked the MTP on trunk in the CUCM. I have never seen this before with no two-way audio between CUC and CUCM.
I'm just wondering if any one out there could explain this or if there is another way to accomplish this without checking the MTP. Thank you!
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