Dial-Peers Required for Calls to FXS Port?

[Matthew Martin]

Hello All,

We have ported our remote branch offices' telephone numbers to our ITSP for our HQ location to now come in to us over our SIP Trunk. Now, that the telephone numbers are coming in to our SIP Trunk I need to setup routing to send their Fax lines to the FXS port of their local Gateway.

I have already configured a Dial-Peer on the SIP CUBE in our HQ location to send the Fax calls directly to each location's Gateway (*ISR4321). All of their other calls get sent to CUCM and I am routing them from there. So the Fax is the only thing I'm having trouble with. However, some, or maybe even most of the faxes are completing successfully. But, I've noticed that the dial-peers being used on their local Gateways are not what I intended, and dial-peer 0, is being used as the inbound dial-peer.

Inbound Routing:

PSTN  --->  SIP CUBE  --->  ISR4321  --->  FXS Port  --->  Fax Machine

*The outbound routing is the same thing, just reversed...

I have T.38 faxing configured under the voice service voip section of the config. And I created a Trunkgroup called "FAX" to direct the call to the FXS port. However, I think I have the Peers reversed. Currently, I have what I thought was an inbound peer being used as the Outbound DIal-Peer when I view"show voice call status" output, and dial-peer 0 being used as the inbound peer. It does seem like T.38 is working because when the call starts show voice call status shows g729r8, then g711ulaw, and then finally it'll show a baud rate like 14400 for example

JWP-4321-SAN# sh voice call st
CallID     CID  ccVdb      Port        Slot/Bay/DSP:Ch  Called #   Codec    MLPP Dial-peers
0x1BE      3678 0x7F3330EECA90 0/2/0            2/1:1  *5551234567 14400    0/4373
1 active call found

Configuration Below:

!
trunk group FAX
!
!...........
!
voice service voip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
 sip
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
!
!
voice translation-rule 2
 rule 1 /^9\([2-9]\)/ /91\1/
!
!
voice translation-profile VOIP-OUT
translate called 2
!
!.............
!
voice-port 0/2/0
 trunk-group FAX
 translate called 2
 no vad
 no comfort-noise
 description Fax Line - 555-123-4567
 station-id name Fax Line
 station-id number 5551234567
 caller-id enable
!
!.............
!
!
dial-peer voice 4373 pots
 trunkgroup FAX
 description Inbound Dial-Peer for Fax Machine - 555-123-4567
 destination-pattern 551234567
!
dial-peer voice 43731 voip
 description Outbound Dial-Peer for Fax Machine - 555-123-4567
 translation-profile outgoing VOIP-OUT
 destination-pattern 91[2-9]..[2-9]......
 session protocol sipv2
 session target ipv4:
 voice-class sip bind control source-interface GigabitEthernet0/0/1.2
 voice-class sip bind media source-interface GigabitEthernet0/0/1.2
 dtmf-relay rtp-nte
 codec g711ulaw
 fax-relay ecm disable
 fax-relay sg3-to-g3
 fax rate 14400
 clid network-number 5551234567
 no vad
!

So I'm thinking I need a VOIP peer for inbound and a POTS peer for outbound. But, I'm not positive what are the necessary commands for each peer, like which one gets destination-pattern, or incoming called-number, etc...?

If anyone has any thoughts or suggestions it would be greatly appreciated. Or if someone could provide example dial-peers for the purposes I described above.

Thanks in Advance,
Matt

[Chris Deren]

Why are you attempting to route directly from CUBE to analog GW and not through CUCM?

Routing thru CUCM provides ability to track these calls in CDR, apply CAC, internal calling between them, etc. I would recommend routing all calls to CUCM and then depending on what protocol controls the analog GWs build it out accordingly, i.e. SIP trunk to analog GW and route patterns matching the analog DNs pointing to the RL/RG/Analog GW trunk. Or SCCP, MGCP port built out in CUCM if that is your protocol of choice.

[Matthew Martin]

I did it this way because I was told by TAC on a few different occasions that when dealing with T.38 Faxing that it is easier to route the calls this way, bypassing CUCM altogether. I'm guessing because there are way less things to configure and less chance for something to go wrong with the fax.

-Matt

[Henrique Ferreira]

hi

were you able to fix it?

[Ishan Zutshi]

your voip dial peer 43731 is going to act as an outgoing dial peer only because it has destination pattern configured
you need to add "incoming called-number XXX" so that your voip dial peer gets matched as inbound dial peer.

something like this:
!
dial-peer voice 43731 voip
incoming called-number .
!

[csrlima]

Hi, did you fix this? im interesting on it becouse i need to use an analog phone with VG202 ( analog voice gateway) but without CUCM : analog phone (pots)-> VG202 -> SIP -> Cisco Cube -> PSTN

I already receive calls from pstn (incoming) but i cant make outbound calls becouse phone gets busy after first digit.

I receive dial tone, i dial 5 numbers or nine, but after first digit ( that arrives to vg202) i got busy tone. Can you help me?

[Ishan Zutshi]

Can you post the config of your:

1) incoming dial peer pots

2) outgoing dial peer voip

3) config of your voice port where the phone is connected?

4) any other relevant config for this call like translation rules etc?