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UC560 CME/CUE no Audio

talhazubairi
Level 1
Level 1

Hi All,

                  

In reference to the subject of this email, I would like to open a case for a solution we have given to one of our Customer.

We are in a pre-implementation phase and trying to simulate the same as needed by the customer.

Call Flow 1:

PBX ---VG------WAN-----CME---IP Phone

Call Flow 2:

PBX ---VG------WAN-----CME/CUE (AA)---IP Phone

Requirement:

Extend Analog lines through a Voice Gateway (VG) over a Wireless Point-to-Point to their Warehouse. No Provider can give Analog lines to Site-B.

Solution:

Telephone Lines are connected on the VG FXO port, which have a 'connection plar' for Site-B Auto Attendent number '4000'. A dial peer is then created on the CME located at Site-B which send it to the AA on the CUE. During this transition I can hear the greeting message and transfer voice, but once the call is forwarded to the required extension, the line from head office drops but I can still see the voice session up at Site-B.

I need to know if this solution is possible or not? If Yes, how is it possible? any example would be very helpful.

I have checked everything and every possible solution over the internet. Transcoding and Codecs are working fine. No Access List, no security.

Thanks & Regards,

TZ

Message was edited by: Talha Zubairi

2 Replies 2

brmeade
Level 4
Level 4

Did you collect any debugs?  Packet captures?  What protocols are you using for your dial-peers?

Hi Brian,

Thanks for your reply. Yes i did collect the debugs (find attached in previous post) also with the relevent configuration. The protocols i used were, between the Wireless PTP "g729r8" and "g711ullaw" with Unity Express. I have also tried "g711" all the way from SiteA to SiteB, but with no success.

SiteA

dial-peer voice 100 voip

destination-pattern 4000

session target ipv4:10.10.10.254

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

 

Site-B

sccp local Vlan10
sccp ccm x.x.x.x identifier 1 version 7.0
sccp ip precedence 3
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register transcode

!
dspfarm profile 1 transcode 
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
maximum sessions 6
associate application SCCP
!

dial-peer voice 1000 voip

description **** Unity Express Auto Attendant***

destination-pattern 4000

session protocol sipv2

session target ipv4:10.10.10.253

dtmf-relay sip-notify

codec g711ulaw

no vad

!

telephony-service

sdspfarm units 6

sdspfarm transcode sessions 6

sdspfarm tag 1 transcode

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