HI all
CUCM-------------WAN-------VOIP-------------------------CME-----Pots
Above is my Scenario where I have 2 Remote site using CME and HQ CUCM 7. For toll bypass I am using SIP Trunk to route traffic from HQ to Branch to make local call from Branches. 1 of 2 sites its working ok , But the Second Site although the calls are routing via SIP they unable to go out to POTS dial peer.
I have tried to compare the configs .the only difference from the workin site is that on the sip conf I have extracted the following, Site A does not Support the following commands.
NOT Supported
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
( I have enclosed Debug from CME not work)
Could this be the problem
NOT WORKING - SITE A
voice service voip
clid network-provided
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
--More-- sip
registrar server
asymmetric payload full
!
!
voice class codec 1
codec preference 1 g729br8
codec preference 2 g711ulaw
codec preference 3 g711alaw
!
!
!
sip-ua
!
!
WORKING - SITE B
voice service voip
clid network-provided
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
registrar server
asymmetric payload full
!
!
voice class codec 1
codec preference 1 g729br8
codec preference 2 g711ulaw
codec preference 3 g711alaw
!
!
