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matthubach
Beginner

Incoming FAX via SIP Trunk to an FXS port on an H323 GW

Can anyone let me know how to get this to work? I have read the message boards and have attempted multiple configurations (MGCP, STCAPP), but I cannot get the incoming fax to work. I haven't bothered with outbound yet.

What I have is a fax number that I recently ported to a SIP provider. The call comes in and I can ring the FXS port, but the fax never comes through.

The way I left (open to ALL suggestions though) it right now is; the incoming fax hits the CUBE (Denver), is routed to Call Manager 7.1 (Denver), and the call manager uses a Xlate Pattern / Route Pattern to send it to the H323 gateway  (Pittsburgh).

1) Is there a better way?

2) If not, can I see someone's dial-peers/config who have this working by chance?

3) FYI...there is no VG224 involved here.

Thanks.

5 REPLIES 5

On your H323 gateway I assume you have FXS cards on them?. What fax protocol are you using? What fax protocol does your ITSP provider support?

Can you send you configuration (sh run)

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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Yes, I have 4 FXS ports.

Trying to set this up using the t38 fax protocol and provider does support T38. I am not getting a lot of help from them so far.

***********************************CUBE******************************************************

dial-peer voice 7000 voip

*** FROM THE CUBE ***

preference 1

destination-pattern [23456789].........

session protocol sipv2

session target ipv4:10.255.201.21 (call manager IP - v 7.1)

session transport udp

incoming called-number [23456789].........

voice-class codec 2

dtmf-relay rtp-nte

no vad

!

voice service voip

address-hiding

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none

h323    

  emptycapability

sip     

  registrar server expires max 3600 min 600

  early-offer forced

  midcall-signaling passthru

  g729 annexb-all

!        

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

!        

voice class codec 2

codec preference 1 g729r8

codec preference 2 g711ulaw

!        

***********************************H323******************************************************

!

dial-peer voice 401 pots

destination-pattern 4231

port 0/2/1

!

dial-peer voice 3000 voip

*** H3232 Gateway ***

progress_ind setup enable 3

session target ipv4:10.255.201.21

incoming called-number 423[01]

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

no vad

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729br8

codec preference 3 g729r8

!

!

!

!

voice class h323 1

  h225 timeout tcp establish 3

  h225 timeout setup 2

h225 display-ie ccm-compatible

  telephony-service ccm-compatible

!

Can you send the ff debugs from the CUBE

1. debug ccsip messages

2. debug voip vtsp all

3. debug fax relay t30 all-level-1

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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Attached are the log outputs.

303-220-0100 faxed to 412-243-9030.

Hi,

I have looked at your trace and there is something that doesnt look quite right during the call setup. Have you edited the IP before sending the trace over..?? In the trace below when cucm sends a 200 ok, it should send the IP address of the h323 gateway where the fax machine resides. However I see the IP address of the CUBE.

Received:
SIP/2.0 200 OK
Date: Fri, 24 Aug 2012 17:05:38 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "CIBER, INC." <>3032200100@siptrunk.myvtel.com>;tag=1E03E668-13F4
Allow-Events: presence, kpml
Supported: replaces
Supported: Geolocation
Remote-Party-ID: <4231>;party=called;screen=no;privacy=off
Content-Length: 237
Require:  timer
To: <4122439030>;tag=b4eca503-64ba-4efd-ac07-76a365113a0c-30948365
Contact: <4122439030>
Content-Type: application/sdp
Call-ID: D2F8C381-ED4011E1-9B8495D9-6809FCA@10.255.205.22
Via: SIP/2.0/UDP 10.255.205.22:5060;branch=z9hG4bK3BFA71401
CSeq: 101 INVITE
P-Preferred-Identity: <4231>
Session-Expires:  1800;refresher=uas

v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 10.255.201.21
s=SIP Call
c=IN IP4 10.255.205.22 (this shuld be the ip add where the fax machine resides)
t=0 0
m=audio 16964 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

1230500: *Aug 24 10:37:37: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:4122439030@10.255.201.21:5060 SIP/2.0
Via: SIP/2.0/UDP 10.255.205.22:5060;branch=z9hG4bK3BFA85D0
From: "CIBER, INC." <>3032200100@siptrunk.myvtel.com>;tag=1E03E668-13F4
To: <4122439030>;tag=b4eca503-64ba-4efd-ac07-76a365113a0c-30948365
Date: Fri, 24 Aug 2012 16:37:27 GMT
Call-ID: D2F8C381-ED4011E1-9B8495D9-6809FCA@10.255.205.22
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

Can you please do another test call and send me the ff debugs from the h323 gateway

debug voip ccapi inout

debug voip vtsp all

debug fax relay t30 all-level-1

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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