We are planning on converting one of our PRI customers to SIP using a TA908 and would like to know if my plan will work. The customer's PBX can only accept a PRI otherwise we would use SIP all the way to their PBX.
Below is our current setup:
Metaswitch -> PRI at CPE -> Cisco PBX -> IP Phones
Here is what we want to do:
Metaswitch -> SIP Into TA908 -> PRI Out to CPE -> Cisco PBX -> IP Phones
We basically are taking the TA and handing off the PRI from there instead of the Metaswitch doing it. Do we build the voice users on the TA908, even though the IP phones will continue to register with the Cisco PBX? The reason we don't take full control of all equipment is because they have a managed services company managing their phones, we only provide the PRI.
Luke,
Nope, in a typical SIP-to-PRI configuration (SIP from Meta and PRI to CPE) you will not configure voice user accounts at all. See link to sample configuration below:
We are testing this in the lab and having some issues getting it working.
Here is our lab setup:
Meta -> TA924 (SIP in, PRI out) -> TA908 (PRI In) -> POTS
We can see the call hit the first TA, but it doesn't seem to be exiting the first TA. I'm sure it has something to do with our config.
Luke,
Is the T1 connection between the TA 924 and 908e in an "UP" state, (via the "show interface t1 0/x") CLI command? If so, is the PRI D-channel between the TA 924 and 908e in an "UP" state (via the "show interface pri x" ) command? Lastly, verify the PRI interface of the TA 924 is configured as "Network Role" and the PRI Interface of the TA 908e is configured as "User Role".
-Craig
Yes to all those questions.
And you have a voice grouped-trunk set up for both of your voice trunks? Also, it would be easier to diagnose any config issues if you could provide a sample of the 924 config, minus any private information.
voice grouped-trunk PRI
trunk T02
accept $
!
voice grouped-trunk SIP
trunk T01
accept $
Below is the full voice config for the TA924. Also note, that calls are traversing to the 908 over the PRI, so I think we are set up correctly on that front.
I'm just not sure how to configure the 908 to test it.
We have a SIP Binding on the metaswitch that the PBX uses. The 924 is set up for SIP to PRI conversion and hands off to the 908. I'd like to test it, but all I get is busy either way. I'm sure if I had a PBX to hand off to it would be easier, but I don't have one readily available.
NG-TA924-LAB#show run voice
Building configuration...
!
!
voice feature-mode network
voice flashhook mode transparent
voice forward-mode network
voice conferencing-mode local
!
!
!
!
!
!
!
!
voice dial-plan 1 local NXX-XXXX
voice dial-plan 2 local 515-NXX-XXXX
voice dial-plan 3 long-distance 1-NXX-NXX-XXXX
voice dial-plan 4 toll-free 1-800-NXX-XXXX
voice dial-plan 5 toll-free 1-888-NXX-XXXX
voice dial-plan 6 toll-free 1-877-NXX-XXXX
voice dial-plan 7 toll-free 1-866-NXX-XXXX
voice dial-plan 8 international 011-$
voice dial-plan 9 specify-carrier 411
!
!
!
!
!
!
voice trunk T01 type sip
sip-server primary x.x.x.x
!
voice trunk T10 type sip
description "SIP-PBX"
match ani "$" add p-asserted-identity "515369xxxx"
sip-server primary x.x.x.x
max-number-calls 23
register 5153xxxxxx auth-name "5153xxxxxx" password encrypted "262cb11554b83d99b9db1a04ce18e9162d35"
!
voice trunk T11 type isdn
description "ISDN-PRI"
resource-selection circular descending
connect isdn-group 1
early-cut-through
rtp delay-mode adaptive
!
!
voice grouped-trunk ISDN
trunk T11
accept $ cost 0
!
!
voice grouped-trunk SIP
trunk T10
accept $ cost 0
!
!
!
!
!
!
!
!
!
!
end
Bump
Anybody else have any advice on this?
Do you have the media-gateway enabled on the interface for SIP Trunk on the TA 924?
interface eth 0/X
description 908e to Network
ip address x.x.x.x x.x.x.x
ip mtu 1500
no rtp quality-monitoring
media-gateway ip primary
qos-policy out Outbound
no awcp
no shutdown
You said you see the call come in to the TA924 but not go out.