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
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.
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,
Message was edited by: Talha Zubairi
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.
dial-peer voice 100 voip
session target ipv4:10.10.10.254
sccp local Vlan10
sccp ccm x.x.x.x identifier 1 version 7.0
sccp ip precedence 3
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register transcode
dspfarm profile 1 transcode
maximum sessions 6
associate application SCCP
dial-peer voice 1000 voip
description **** Unity Express Auto Attendant***
session protocol sipv2
session target ipv4:10.10.10.253
sdspfarm units 6
sdspfarm transcode sessions 6
sdspfarm tag 1 transcode