02-17-2014 04:13 AM - edited 03-16-2019 09:46 PM
Hi there,
We are facing a voice quality probem in our voip solution using CUCM 8.6 and AS5400 GWs;
Some PSTN calls using PRI lines are experiencing very bad quality.(1 call per 50 more or less)
The called number can be mobile or landline, and i can know that the call will have this problem even before the call is established, just hearing the ringback tone (Distorsion). Sometimes i can also hear third party conversation mixed up with my call .
All the serial interfaces and controllers are error free.
The same problem when i use a second AS5400 located on a different datacenter.
The architecture is as follow:
CUCM<-->SIP Trunk<-->AS5400<-->PRI<-->PSTN
Has anyone else faced the same issue /sympthoms?
Thanks in advance
02-17-2014 04:35 AM
Hi Youseef,
Some inputs and workaround for the same is as below:-
One more approach to confirm if there is problem with particular channel then make some directed or test calls for that particular channel in such a way that the call occupies that faulty channel.
Once you confirm that the call distortion happens only for particular channel then ask your PSTN to reset that channels.
Regards,
Nishant Savalia
02-17-2014 04:41 AM
Hi Nishant,
Thank you very much for your answer:
1) The voice quality problem is inbound and outbound.
2)This is happening even when no calls are active on the PRI and AS5400.
3) We have 4 PRI and i already tested them all, the problems appears in different channels on all the PRIs
4)How can i block or route calls to specific channel into a PRI?
Regards
02-17-2014 06:43 AM
Hi,
2)This is happening even when no calls are active on the PRI and AS5400.
Can you please elaborate the above line
4)How can i block or route calls to specific channel into a PRI?
To block the calls, please refer below link
https://supportforums.cisco.com/docs/DOC-4989
To route the call through specific channel is usuall a tedious work when you have multiple channels to test.
It's time consuming so to avoid it you can ask your PSTN to do such test.
But still you can test as below:-
Following command needs to be issued on the Gateway:
controller E1 0/1/0
trunk-group first timeslots 1-13
trunk-group sec timeslots 14-15
!
dial-peer voice XXXX pots
trunkgroup sec
destination-pattern XXXX
! --- Create POTS dial-peer and apply trunk group which contains your desired channel
Regards,
Nishant Savalia
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