I'm involved in the following setup illustrated below. I am Site 4. I have a SIP trunk via the CUBE to Site 1. Calls route between Site 4 and Site 1 with no problem. Is it possible to use Site 1 CUCM, to route calls from Site 4 to Sites 3 or 2? Is it possible for a user from site 4 to call a user in site 2, using Site 1 as call route point? If so, what needs to be setup? Right now, on Site 4 CUBE, I receive a SIP/2.0 404 Not Found from the Site 1 CUCM. Mind you, I don't own Site 1, 2 or 3. But, I can make recommendations to them.
Ultimately I'm trying to avoid making additional SIP trunks to my CUBE.
I would recommend you to put in these changes as with the configuration you have outlined above you open up for call loops, or at least hairpin calls, as the .T destination would match on none existing directory numbers in the 444.... range if your (Site 4) sends a 404 Not Found to your Cube.
voice class sip-options-keepalive 1
description Used for Site 1 SIP OPTIONS PING
!
voice class sip-options-keepalive 4
description Used for Site 4 SIP OPTIONS PING
!
voice class dpg 1
dial-peer 2000
!
voice class dpg 4
dial-peer 1000
!
voice class e164-pattern-map 1
description E164 Pattern Map for called number to Site 1
e164 1.BAD !Not actually used for call routing as that is done by DPG 1, but needed for the dial-peer to be in operational state
!
voice class e164-pattern-map 4
description E164 Pattern Map for called number to Site 4
e164 4.BAD !Not actually used for call routing as that is done by DPG 4, but needed for the dial-peer to be in operational state
!
voice class uri SITE1 sip
host ipv4:<site 1 CM IP> !Add more lines is needed
!
voice class uri SITE4 sip
host ipv4:<site 4 CM IP> !Add more lines is needed
!
dial-peer voice 1001 voip
desc Inbound CUCM to CUBE from Site 4
session protocol sipv2
destination dpg 1 !Use DPG to send calls to site 1
no incoming called-number .T
incoming uri via SITE4 !Use information in the VIA header to match the inbound calls
no voice-class sip options-keepalive !Not needed on inbound dial-peers
voice-class sip bind control source-interface fastethernet0/0
voice-class sip bind media source-interface fastethernet0/0
codec g711ulaw !It would be better likely if you where to have a voice-class codec list for this
no vad
!
dial-peer voice 1000 voip
desc Outbound CUBE to CUCM to Site 4
session protocol sipv2
no destination-pattern 444....
destination e164-pattern-map 4 !Define the destination number with a pattern map instead of having individual patterns on dial-peers
session-target ipv4:(CUCM4 IP)
no voice-class sip options-keepalive
voice-class sip options-keepalive profile 1 !Use the more modern way of defining SIP option ping
voice-class sip bind control source-interface fastethernet0/0
voice-class sip bind media source-interface fastethernet0/0
codec g711ulaw
no vad
!
dial-peer voice 2001 voip
desc Inbound CUCM to CUBE from Site 1
session protocol sipv2
destination dpg 4 !Use DPG to send calls to site 4
no incoming called-number 444....
incoming uri via SITE1 !Use information in the VIA header to match the inbound calls
no voice-class sip options-keepalive !Not needed on inbound dial-peers
voice-class sip bind control source-interface fastethernet0/1
voice-class sip bind media source-interface fastethernet0/1
codec g711ulaw !It would be better likely if you where to have a voice-class codec list for this
no vad
!
dial-peer voice 2000 voip
desc Outbound CUBE to CUCM to Site 1
session protocol sipv2
no destination-pattern .T
destination e164-pattern-map 1 !Define the destination number with a pattern map instead of having individual patterns on dial-peers
session-target ipv4:(CUCM1 IP)
no voice-class sip options-keepalive
voice-class sip options-keepalive profile 4 !Use the more modern way of defining SIP option ping
voice-class sip bind control source-interface fastethernet0/1
voice-class sip bind media source-interface fastethernet0/1
codec g711ulaw !It would be better likely if you where to have a voice-class codec list for this
no vad
If you use a codec list add this
voice class codec 1
codec preference 1 g722-64
codec preference 2 g711alaw
codec preference 3 g711ulaw
!
And on each dial-peer do this
no codec g711ulaw
voice-class codec 1
My suggested configuration uses dial peer groups to define the egress dial-peer for each call direction as all that you want to achieve is that any call from site 1 goes to site 4 and the reverse where all calls from site 4 goes to site 1. You can see DPGs as a way to hard code the path to use and not utilize the actual called numbers to do this.
