Configuration help with PRI ISDN routing

[oahuoahu1]

I am new to voice configuration and need some help with my config.   I do have PVDM modules installed on the NIM-MFT cards.

Avaya PBX ---T1 pri---- 4331 w/mft t1 card --------MPLS---------4331 w/mft t1 card ------T1 pri-------PBX

I have the T1 pri's up and it appears that the PBX is up, but the calls do not go through.  I need help to see what I am missing.

Thank you for any help.

[Nithin Eluvathingal]

What u see on debug isdn

and what’s the status of isdn

[oahuoahu1]

I have added the log file, but I have changed the numbers a bit.

[Roger Kallberg]

Can you please provide a bit more details on the call flow with number information included for the two systems at each end of this setup? You’re dial peers are not very well defined, those need to be worked on a bit to make them more specific. But without any insight into the specifics of the setup you have it’s pretty near impossible to give any advice of value.

[oahuoahu1]

I have added the log file.  I have changed the numbers a bit.

[Roger Kallberg]

Would you mind to write down information about called and calling number on both ends so that we don't need to dig that up from the log file.

[oahuoahu1]

Called 5559129 (off of router 2 side pbx) from 5552254579 (originating from the router 1 side pbx).  Both PBXs are State = MULTIPLE_FRAME_ESTABLISHED, but the call doesn't go from router 1 to router 2.

[Roger Kallberg]

Can you please outline what the ranges of numbers are on each side of the call path?

[oahuoahu1]

I do not know that information.  This is part of the defense switched network.  There are two sites, one with a Nortel MSL-100 and the other site has a Avaya 8710 media gateway.  Each of these devices connect to a Cisco 4331 router with a NIM-MFT2-T1/E1 card with PVDM4-32 module at each site with a T1 PRI line.  The 4331s are connected to a MPLS link.  I need to find the best way to have the calls route to and from each other side.  I only have access to the 4331s.   In short...I just need to patch the sites together (if that makes sense).  I do not have much experience on the voice side of things and do not know enough to ask the necessary questions.  I do appreciate the help.

[Roger Kallberg · Accepted solution]

With the given information it’s not possible to give you a very specific answer, but something in line with this should give you something to start with.

In both GW
voice class codec 1
 codec preference 1 g711ulaw

Calling PBX GW, aka originating side of call
dial-peer voice 1 pots
 description Inbound dial peer from Avaya PBX GW
 direct-inward-dial
 incoming called-number .
!
dial-peer voice 10 pots
 description Outbound dial peer to Avaya PBX GW
 destination-pattern 5552254579$  or 5552......$ if that works for you to not have an overlap
 port 1/0/0:23 not sure I got the port correct, please verify
 no digit-strip
!
dial-peer voice 99 voip
 description Inbound dial peer from PBX GW
 voice-class codec 1
 session protocol sipv2
 incoming called-number .
 dtmf-relay rtp-nte sip-kpml
 no vad
!
dial-peer voice 999 voip
 description Outbound dial peer to PBX GW
 destination-pattern 5559129$ or 5559...$ if that works for you to not have an overlap
 voice-class codec 1
 session protocol sipv2
 session target ipv4:<IP of called GW>
 supplementary-service pass-through
 dtmf-relay rtp-nte sip-kpml
 no vad

Called PBX GW, aka receiving side of call
dial-peer voice 1 pots
 description Inbound dial peer from PBX GW
 direct-inward-dial
 incoming called-number .
!
dial-peer voice 10 pots
 description Outbound dial peer to PBX GW
 destination-pattern 5559129$ or 5559...$ if that works for you to not have an overlap
 port 0/1/0:23 not sure I got the port correct, please verify
 no digit-strip
!
dial-peer voice 99 voip
 description Inbound dial peer from Avaya PBX GW
 voice-class codec 1
 session protocol sipv2
 incoming called-number .
 dtmf-relay rtp-nte sip-kpml
 no vad
!
dial-peer voice 999 voip
 description Outbound dial peer to Avaya PBX GW
 destination-pattern 5552254579$  or 5552......$ if that works for you to not have an overlap
 voice-class codec 1
 session protocol sipv2
 session target ipv4:<IP of calling GW>
 supplementary-service pass-through
 dtmf-relay rtp-nte sip-kpml
 no vad
!

Very likely you would need to adopt this and possibly also translate the called and calling numbers to fit the format on each side.

[oahuoahu1]

Thank you.  Since every call that comes into one of the routers is always going to the other side can I just use a wildcard to route all calls to the other router?  something like destination-pattern .$

[Roger Kallberg]

No that’s why it didn’t work for you before. My example is for sure very specific, that’s why I asked you for range information. You can not use my example as is, you’ll have to read up on dial peer operation and adopt it for your own needs.

What you need to achieve is to have the outbound dial peers, those are the once with either session target or a port on them, be none overlapping. That way the router can determine the direction of the call. With the match all wild card match that you have in your configuration the router would not know where to send the call.

I did not review your debug output, as with that configuration in place it seemed to not be worthwhile. If you don’t get it to work I suggest that you run a deb voip ccapi inout and deb isdn q931 to figure out what happens.

[Roger Kallberg]

Recommend you to read this excellent document. https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html

[oahuoahu1]

Thank you for all your help.  I will modify to our needs and try this early next week. 

[Roger Kallberg]

I reworked my original reply to be easier to read and to better describe the called and calling sides in a clear way.