Dear Team,
I have an SRST scenario for my customer that have SIP and SCCP phones in its Branches. Branches are connected to HQ CUCM using
MulitSite Centralized IPT enviroment and the branches gateways are added as an H.323 gateways.
as we are testing the SRST feature the SCCP phones (8945) are fallover to the SRST gateway successfully as well as the SIP phones (7841).
SCCP phones are being able to call each other and to call SIP Phones.
However SIP phones are not being able to call any SIP,SCCP IP phones.
while debugging the VGW (attatched) along with its show run the following occurs:
1. SCCP to SIP --> a dial peer is matched and the call goes through.
2. SIP to SCCP --> No Dial Peer is matched and the call does not go through.
3. SIP to SIP --> No Dial Peer is matched and the call does not go through
The below is the configuration for SIP
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
bind control source-interface GigabitEthernet0/0.31
bind media source-interface GigabitEthernet0/0.31
registrar server expires max 200 min 60
!
!
voice register global
mode srst
system message Customer
max-dn 10
max-pool 10
dialplan-pattern 1 7... extension-length 4
application sip
!
voice register pool 5
translation-profile outgoing emergency
id network 172.20.71.0 mask 255.255.255.0
no digit collect kpml
dtmf-relay sip-notify
codec g711ulaw
no vad
SCCP Config
call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 172.20.31.222 port 2000
max-ephones 25
max-dn 25
system message primary Customer_SRST
Can you please help me why i am not being able to do Outbound Dialing using the SIP phones? and is there any automated dial-peers that
need to be generated i am being missing in the configuration.
Thanks
Ammar Said
