12-30-2013 06:33 AM - edited 03-16-2019 09:02 PM
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
12-30-2013 07:09 AM
Did you collect any debugs? Packet captures? What protocols are you using for your dial-peers?
12-30-2013 10:50 PM
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
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide