I am trying to config a 908e to receive SIP on the ingress Trunk T01 from the service provider. Trunk T02 is PRI to PBX. As well as 3 FXS lines. I have researched the forums and think I may be close. Will this configuration work?
!
timing-source internal
!
timing-source internal secondary
!
!
interface eth 0/1
description NOT_IN_USE
no ip address
shutdown
!
interface eth 0/2
description HSTSTXSIN00_Z02
ip address 10.56.x.x 255.255.255.248
media-gateway ip primary
no shutdown
!
!
interface t1 0/1
shutdown
!
interface t1 0/2
shutdown
!
interface t1 0/3
description TestPRI#1
tdm-group 1 timeslots 1-24 speed 64
no shutdown
!
interface t1 0/4
shutdown
!
Voice user 2145551212
connect fxs 0/1
first-name Test
last-name Network
description Line #1
caller-id-override external-name Test
caller-id-override external-number 2145551212
did 2145551212
no shutdown
!
voice user 2145551213
connect fxs 0/2
first-name Test
last-name Network
description Line #2
caller-id-override external-name Test
caller-id-override external-number 2145551213
did 2145551213
no shutdown
!
voice user 2145551214
connect fxs 0/3
first-name Test
last-name Network
description Line #3
caller-id-override external-name Test
caller-id-override external-number 2145551214
did 2145551214
no shutdown
!
interface fxs 0/4
shutdown
!
interface fxs 0/5
shutdown
!
interface fxs 0/6
shutdown
!
interface fxs 0/7
shutdown
!
interface fxs 0/8
shutdown
!
interface fxo 0/0
shutdown
!
!
interface pri 1
isdn name-delivery setup
calling-party override if-no-CID
calling-party number 2145551001
calling-party name-facility-timeout 0
connect t1 0/3 tdm-group 1
role network b-channel-restarts disable
digits-transferred 10
no shutdown
!
isdn-group 1
connect pri 1
!
voice feature-mode network
voice forward-mode network
!
!
voice codec-list siptrunk
codec g729
codec g711ulaw
!
!
voice trunk T01 type sip
description "T01"
sip-server primary sip-trk01.ISP.net
codec-group siptrunk
no reject-external
default-ring-cadence internal
!
voice trunk T02 type isdn
resource-selection linear ascending
connect isdn-group 1
modem-passthrough
rtp delay-mode adaptive
codec-group siptrunk
no reject-external
!
!
voice grouped-trunk MAIN
description "SIP to ISP"
trunk T01
accept 214-xxx-xxxx cost 0
accept 469-xxx-xxxx cost 0
accept 972-xxx-xxxx cost 0
accept 1-Nxx-Nxx-xxxx cost 0
accept 1-800-Nxx-xxxx cost 0
accept 1-888-Nxx-xxxx cost 0
accept 1-877-Nxx-xxxx cost 0
accept 1-866-Nxx-xxxx cost 0
accept 1-855-Nxx-xxxx cost 0
accept 011-$ cost 0
accept 411 cost 0
accept 611 cost 0
accept 911 cost 0
accept 0-Nxx-Nxx-xxxx cost 0
accept $ cost 0
reject 214-555-1212 cost 0
reject 214-555-1213 cost 0
reject 214-555-1214 cost 0
!
voice grouped-trunk PRI
description "To Customer PBX"
trunk T02
accept $ cost 0
reject 214-555-1212 cost 0
reject 214-555-1213 cost 0
reject 214-555-1214 cost 0
!
!
voice dial-plan 1 local 214-555-1212
voice dial-plan 2 local 214-555-1213
voice dial-plan 3 local 214-555-1214
voice dial-plan 4 local NXX-NXX-XXXX
voice dial-plan 5 long-distance 1-NXX-NXX-XXXX
voice dial-plan 6 international 011-$
!
!
ip route 0.0.0.0 0.0.0.0 x.x.x.x
!
no ip tftp server
no ip tftp server overwrite
no ip http server
ip http secure-server
no ip snmp agent
no ip ftp server
no ip scp server
no ip sntp server
!
ip sip
ip sip udp 5060
no ip sip tcp
!
no ip sip registrar authenticate
ip sip registrar realm metaswitch.ISP.net
!
end
wr mem
Since all your users are directly connected to the PBX you do not need a grouped-trunk. Your grouped-trunk to the outside world also has reject statements that don't need to be there, and accept statements that can be summarized better. You should remove the grouped-trunk PRI entirely and your MAIN should look like this IMO:
voice grouped-trunk MAIN
description "SIP to ISP"
trunk T01
accept Nxx-Nxx-xxxx cost 0
accept 1-Nxx-Nxx-xxxx cost 0
accept 1-800-Nxx-xxxx cost 0
accept 1-888-Nxx-xxxx cost 0
accept 1-877-Nxx-xxxx cost 0
accept 1-866-Nxx-xxxx cost 0
accept 1-855-Nxx-xxxx cost 0
accept 011-$ cost 0
accept 411 cost 0
accept 611 cost 0
accept 911 cost 0
accept 0-Nxx-Nxx-xxxx cost 0
Having an 'accept $ cost 0' statement means that it'll accept any digits the user throws at it, including a 50 digit string. You don't want that.
Secondly, since these are analog stations, you need to define a dialplan on the system to define what digit strings are accept able. You should add the following:
voice dial-plan 1 always-permitted [4,6,9]XX
voice dial-plan 2 local nxx-nxx-xxxx
voice dial-plan 3 long-distance 1-nxx-nxx-xxxx
voice dial-plan 4 long-distance 1-8xx-nxx-xxxx
voice dial-plan 5 international 011-$
voice dial-plan 6 international 0-nxx-nxx-xxxx
Hope this helps!
I didn't know you could use a bare trunk like that. Do calls from the SIP (grouped) trunk route to the PRI automatically this way? Could you provide a more complete voice config example?
Chris
Thank you.....i'll give it a try. Didn't know I the trunked-group PRI statement wasn't required.
Did you get this working, I'm trying to do the same thing.
I have taken a working config that does SIP<>PRI only, at another customer site and provisioned the analog ports with an extension. The analog ports work, the problem is I can't test the PRI until we install the unit at the customer site.
Thanks,
Greg
Hi gregstahl:
You might want to start a new thread and post your configuration (remove passwords). We can review it and give you feedback if you like.
Best,
Chris
Thanks Chris. I will when we get closer to the install.
Greg