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.
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
! 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,
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.
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.