SIP to PRI and FXS Config?

[tszala]

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

[phonealchemist]

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!

[Anonymous]

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

[tszala]

Thank you.....i'll give it a try.  Didn't know I the trunked-group PRI statement wasn't required. 

[gregstahl]

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

[Anonymous]

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

[gregstahl]

Thanks Chris.  I will when we get closer to the install.

Greg