T.38 Fax Gateway Asterisk 10-2 Beta

Jansalex · October 29, 2011, 4:24pm

Hi all,

I´m testing the new T38 Fax Gateway feature on Asterisk 10.2 Beta.

I put like is in this page :

exten => 1,1,NoOp()
exten => 1,n,Set(FAXOPT(gateway)=yes)
exten => 1,n,Dial(SIP/mypeer,20)

I put this in outbounds calls and inbound Dialplan.

Wel, if I send a FoIP from my computer ( Via GoldBolt and 3CX ), the fax pass via Asterisk and the fax is sended to a fax machine under analogic line. This works fine.

My problem is to receive fax on the computer. I have a Fax Machine via analog line, I send the fax via asterisk, but when the fax arrive to my PBX client ( 3CX in my case ), the call terminate an noting occurs.

The Dialplan is similar at inbound and outbound calls. Why work in one way and not in the other ?. Is not the same configuration in the diaplplan ?

We need to have faxenable=yes ? or with the gateway this is not the way ?

Something who have this Gateway working ?

Thanks.

cerien.jean · November 16, 2011, 10:21pm

Hi

I have a similar / related issue.

Setup: (fax machine) – (ATA HT503) ----t38 ---- (ast 10) —g711— (ast 1.6) — dahdi card

When i register the fax straight on the 1.6 box, with g711, it succeeds. Now I am trying to have the t38 up & running.

When the fax is sent from the fax machine, T38 negotiation happends (reinvites). However, it fails on the ast 10 rc2 box with the following messages:

[Nov 16 16:43:51] ERROR[22522]: res_fax.c:1009 fax_session_new: Could not locate a FAX technology module with capabilities (GATEWAY) [Nov 16 16:43:51] ERROR[22522]: res_fax.c:2503 fax_gateway_start: Can't create a FAX session, gateway attempt failed. [Nov 16 16:43:51] ERROR[22522]: res_fax.c:2775 fax_gateway_detect_t38: error starting T.38 gateway for T.38 channel SIP/pollux-00000011 and G.711 channel SIP/331001-00000010

I’ve tried to recompile with spandsp & app_fax instead of res_fax, no changes

Any suggestions ?

Rgds,

J.

sshoemaker · March 26, 2013, 9:03pm

I know this thread is terribly old, but I was wondering if anyone got this working. I am currently setting up a PSTN(FXO) —> HT503 —> Fax Machine (FXS) —>Asterisk (WAN)… This works great for voice calls in and out… however I can’t seem to get the Fax machine to ever be able to get a fax… I’m assuming that because I’m handing off all calls after 2 rings to Asterisk via the FXO-to-WAN connection that Asterisk then has to somehow know to answer the fax as a fax separate from a phone call and then pass that fax back to the HT503 on the FXS port.

So far, I have the following in my extensions.conf:

[code][general]
autofallthrough=yes
videosupport=yes
t38pt_udptl = yes

[from-trunk]
exten => ata,1,Answer(500)
same => n,Set(TIMEOUT(digit)=2)
same => n,Wait(1)
same => n(menuprompt),Background(main-menu) ; MAIN MENU PROMPT
same => n,WaitExten(4)[/code]

However in my asterisk messages log, I am getting:

[Mar 26 16:46:09] WARNING[32616][C-0000000b] chan_sip.c: Failed to initialize UDPTL, declining image stream [Mar 26 16:46:09] WARNING[32616][C-0000000b] chan_sip.c: Failing due to no acceptable offer found
which I assume is because I am not answering the fax (due to not knowing how to pick up the fax tones)

On the CLI during the fax call(lightly scrubbed):

[code]
<— SIP read from UDP:192.168.x.x:5062 —>
INVITE sip:ata@192.168.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK849329957;rport
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 30 INVITE
Contact: sip:ata@192.168.x.x:5062
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.1B 1.0.5.5
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 453

v=0
o=ata 8002 8000 IN IP4 192.168.x.x
s=SIP Call
c=IN IP4 192.168.x.x
t=0 0
m=audio 5013 RTP/AVP 0 8 4 18 2 97 102 100 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
<------------->
— (14 headers 20 lines) —
Sending to 192.168.x.x:5062 (no NAT)
Using INVITE request as basis request - 478414669-5062-4@BJC.BGI.B.BEC
Found peer ‘ata’ for ‘14154499998’ from 192.168.x.x:5062

<— Reliably Transmitting (no NAT) to 192.168.x.x:5062 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK849329957;received=192.168.x.x;rport=5062
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as413f971b
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 30 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6be1df7c"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘478414669-5062-4@BJC.BGI.B.BEC’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.x.x:5062 —>
ACK sip:ata@192.168.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK849329957;rport
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as413f971b
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 30 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:192.168.x.x:5062 —>
INVITE sip:ata@192.168.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK1033985457;rport
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 31 INVITE
Contact: sip:ata@192.168.x.x:5062
Authorization: Digest username=“ata”, realm=“asterisk”, nonce=“6be1df7c”, uri=“sip:ata@192.168.x.x:5060”, response=“8817c026dd9425203bfe8906b92a9851”, algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.1B 1.0.5.5
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 453

v=0
o=ata 8002 8000 IN IP4 192.168.x.x
s=SIP Call
c=IN IP4 192.168.x.x
t=0 0
m=audio 5013 RTP/AVP 0 8 4 18 2 97 102 100 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
<------------->
— (15 headers 20 lines) —
Sending to 192.168.x.x:5062 (no NAT)
Using INVITE request as basis request - 478414669-5062-4@BJC.BGI.B.BEC
Found peer ‘ata’ for ‘14154499998’ from 192.168.x.x:5062
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found unknown media description format G729E for ID 102
Found unknown media description format AAL2-G726-16 for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729|g722), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.x.x:5013
Looking for ata in from-trunk (domain 192.168.x.x)
list_route: hop: sip:ata@192.168.x.x:5062

<— Transmitting (no NAT) to 192.168.x.x:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK1033985457;received=192.168.x.x;rport=5062
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 31 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:ata@192.168.x.x:5060
Content-Length: 0

<------------>
– Executing [ata@from-trunk:1] Answer(“SIP/ata-00000008”, “500”) in new stack
Audio is at 15406
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.x.x:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK1033985457;received=192.168.x.x;rport=5062
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 31 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:ata@192.168.x.x:5060
Content-Type: application/sdp
Content-Length: 329

v=0
o=root 323402420 323402420 IN IP4 192.168.x.x
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.x.x
t=0 0
m=audio 15406 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<— SIP read from UDP:192.168.x.x:5062 —>
ACK sip:ata@192.168.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK677300166;rport
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 31 ACK
Contact: sip:ata@192.168.x.x:5062
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.1B 1.0.5.5
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
– Executing [ata@from-trunk:2] Set(“SIP/ata-00000008”, “TIMEOUT(digit)=2”) in new stack
– Digit timeout set to 2.000
– Executing [ata@from-trunk:3] Wait(“SIP/ata-00000008”, “1”) in new stack
– Executing [ata@from-trunk:4] BackGround(“SIP/ata-00000008”, “main-menu”) in new stack
– <SIP/ata-00000008> Playing ‘main-menu.slin’ (language ‘en’)

<— SIP read from UDP:192.168.x.x:5062 —>
INVITE sip:ata@192.168.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK1602593586;rport
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 32 INVITE
Contact: sip:ata@192.168.x.x:5062
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.1B 1.0.5.5
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 270

v=0
o=ata 8002 8001 IN IP4 192.168.x.x
s=SIP Call
c=IN IP4 192.168.x.x
t=0 0
m=image 5013 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:280
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
— (14 headers 12 lines) —
Sending to 192.168.x.x:5062 (no NAT)
[Mar 26 16:46:09] WARNING[32616][C-0000000b]: chan_sip.c:10277 process_sdp: Failed to initialize UDPTL, declining image stream
[Mar 26 16:46:09] WARNING[32616][C-0000000b]: chan_sip.c:10421 process_sdp: Failing due to no acceptable offer found

<— Reliably Transmitting (no NAT) to 192.168.x.x:5062 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK1602593586;received=192.168.x.x;rport=5062
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 32 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

<— SIP read from UDP:192.168.x.x:5062 —>
ACK sip:ata@192.168.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK1602593586;rport
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 32 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:192.168.x.x:5062 —>
INVITE sip:ata@192.168.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK872483153;rport
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 33 INVITE
Contact: sip:ata@192.168.x.x:5062
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.1B 1.0.5.5
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 272

v=0
o=ata 8002 8002 IN IP4 192.168.x.x
s=SIP Call
c=IN IP4 192.168.x.x
t=0 0
m=audio 5013 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
a=silenceSupp:off - - - -
<------------->
— (14 headers 13 lines) —
Sending to 192.168.x.x:5062 (no NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729|g722), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.x.x:5013

<— Transmitting (no NAT) to 192.168.x.x:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK872483153;received=192.168.x.x;rport=5062
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 33 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:ata@192.168.x.x:5060
Content-Length: 0

<------------>
Audio is at 15406
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.x.x:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK872483153;received=192.168.x.x;rport=5062
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 33 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:ata@192.168.x.x:5060
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 323402420 323402421 IN IP4 192.168.x.x
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.x.x
t=0 0
m=audio 15406 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<— SIP read from UDP:192.168.x.x:5062 —>
ACK sip:ata@192.168.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK505851456;rport
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 33 ACK
Contact: sip:ata@192.168.x.x:5062
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.1B 1.0.5.5
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
— (12 headers 0 lines) —

<— SIP read from UDP:192.168.x.x:5060 —>
SUBSCRIBE sip:*98@192.168.x.x:5060 SIP/2.0
Accept: application/dialog-info+xml
Via: SIP/2.0/UDP 192.168.x.x;branch=z9hG4bK532ce20835bcdd8ec
Route: sip:192.168.x.x:5060;lr
Max-Forwards: 70
From: sip:seth@192.168.x.x:5060;tag=fd7db2268d
To: sip:*98@192.168.x.x:5060
Call-ID: 89aa93b67b79db9d
CSeq: 20345 SUBSCRIBE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“seth”,realm=“asterisk”,nonce=“65b24471”,uri=“sip:*98@192.168.x.x:5060”,response=“030efd24d92698eb309921ed0eda39a2”,algorithm=MD5
Contact: sip:seth@192.168.x.x:5060;transport=udp;+sip.instance="urn:uuid:00000000-0000-1000-8000-00085D394E18"
Event: dialog
Expires: 3597
Supported: path
User-Agent: Aastra 6731i/3.2.2.2104
Content-Length: 0

<------------->
— (18 headers 0 lines) —
Creating new subscription
Sending to 192.168.x.x:5060 (no NAT)
Found peer ‘seth’ for ‘seth’ from 192.168.x.x:5060
Looking for *98 in from-sip-internal (domain 192.168.x.x)

<— Transmitting (no NAT) to 192.168.x.x:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.x.x;branch=z9hG4bK532ce20835bcdd8ec;received=192.168.x.x
From: sip:seth@192.168.x.x:5060;tag=fd7db2268d
To: sip:*98@192.168.x.x:5060;tag=as01fa7e2a
Call-ID: 89aa93b67b79db9d
CSeq: 20345 SUBSCRIBE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘89aa93b67b79db9d’ Method: SUBSCRIBE
– Executing [ata@from-trunk:5] WaitExten(“SIP/ata-00000008”, “4”) in new stack
– Timeout on SIP/ata-00000008, going to ‘t’
– Executing [t@from-trunk:1] WaitExten(“SIP/ata-00000008”, “4”) in new stack
– Timeout on SIP/ata-00000008, continuing…
– Executing [t@from-trunk:2] Goto(“SIP/ata-00000008”, “ata,menuprompt”) in new stack
– Goto (from-trunk,ata,4)
– Executing [ata@from-trunk:4] BackGround(“SIP/ata-00000008”, “main-menu”) in new stack
– <SIP/ata-00000008> Playing ‘main-menu.slin’ (language ‘en’)

<— SIP read from UDP:192.168.x.x:5060 —>
SUBSCRIBE sip:*98@192.168.x.x:5060 SIP/2.0
Accept: application/dialog-info+xml
Via: SIP/2.0/UDP 192.168.x.x;branch=z9hG4bKfa9b55a89565fb660
Route: sip:192.168.x.x:5060;lr
Max-Forwards: 70
From: sip:seth@192.168.x.x:5060;tag=fd7db2268d
To: sip:*98@192.168.x.x:5060
Call-ID: 89aa93b67b79db9d
CSeq: 20346 SUBSCRIBE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“seth”,realm=“asterisk”,nonce=“65b24471”,uri=“sip:*98@192.168.x.x:5060”,response=“030efd24d92698eb309921ed0eda39a2”,algorithm=MD5
Contact: sip:seth@192.168.x.x:5060;transport=udp;+sip.instance="urn:uuid:00000000-0000-1000-8000-00085D394E18"
Event: dialog
Expires: 3597
Supported: path
User-Agent: Aastra 6731i/3.2.2.2104
Content-Length: 0

<------------->
— (18 headers 0 lines) —
Creating new subscription
Sending to 192.168.x.x:5060 (no NAT)
list_route: hop: sip:seth@192.168.x.x:5060;transport=udp
Found peer ‘seth’ for ‘seth’ from 192.168.x.x:5060

<— Transmitting (no NAT) to 192.168.x.x:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.x.x;branch=z9hG4bKfa9b55a89565fb660;received=192.168.x.x
From: sip:seth@192.168.x.x:5060;tag=fd7db2268d
To: sip:*98@192.168.x.x:5060;tag=as5257c9ea
Call-ID: 89aa93b67b79db9d
CSeq: 20346 SUBSCRIBE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="709760d9"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘89aa93b67b79db9d’ in 32000 ms (Method: SUBSCRIBE)
Reliably Transmitting (no NAT) to 192.168.x.x:5062:
OPTIONS sip:ata@192.168.x.x:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5060;branch=z9hG4bK111e03a3
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.x.x;tag=as74cf3198
To: sip:ata@192.168.x.x:5062
Contact: sip:asterisk@192.168.x.x:5060
Call-ID: 0e01db360d8307215a28bbe6016e5225@192.168.x.x:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.1
Date: Tue, 26 Mar 2013 20:46:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.x.x:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.x.x:5060;branch=z9hG4bK111e03a3
From: “asterisk” sip:asterisk@192.168.x.x;tag=as74cf3198
To: sip:ata@192.168.x.x:5062;tag=63705147
Call-ID: 0e01db360d8307215a28bbe6016e5225@192.168.x.x:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.1B 1.0.5.5
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘0e01db360d8307215a28bbe6016e5225@192.168.x.x:5060’ Method: OPTIONS
– Executing [ata@from-trunk:5] WaitExten(“SIP/ata-00000008”, “4”) in new stack
– Timeout on SIP/ata-00000008, going to ‘t’
– Executing [t@from-trunk:1] WaitExten(“SIP/ata-00000008”, “4”) in new stack
– Timeout on SIP/ata-00000008, continuing…
– Executing [t@from-trunk:2] Goto(“SIP/ata-00000008”, “ata,menuprompt”) in new stack
– Goto (from-trunk,ata,4)
– Executing [ata@from-trunk:4] BackGround(“SIP/ata-00000008”, “main-menu”) in new stack
– <SIP/ata-00000008> Playing ‘main-menu.slin’ (language ‘en’)

<— SIP read from UDP:192.168.x.x:5060 —>
SUBSCRIBE sip:*98@192.168.x.x:5060 SIP/2.0
Accept: application/dialog-info+xml
Via: SIP/2.0/UDP 192.168.x.x;branch=z9hG4bK4d6318fcc41af6f29
Route: sip:192.168.x.x:5060;lr
Max-Forwards: 70
From: sip:seth@192.168.x.x:5060;tag=fd7db2268d
To: sip:*98@192.168.x.x:5060
Call-ID: 89aa93b67b79db9d
CSeq: 20347 SUBSCRIBE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“seth”,realm=“asterisk”,nonce=“709760d9”,uri=“sip:*98@192.168.x.x:5060”,response=“11af31cb7cc3c5e3ad1f73b8bbe862ea”,algorithm=MD5
Contact: sip:seth@192.168.x.x:5060;transport=udp;+sip.instance="urn:uuid:00000000-0000-1000-8000-00085D394E18"
Event: dialog
Expires: 3597
Supported: path
User-Agent: Aastra 6731i/3.2.2.2104
Content-Length: 0

<------------->
— (18 headers 0 lines) —
Creating new subscription
Sending to 192.168.x.x:5060 (no NAT)
Found peer ‘seth’ for ‘seth’ from 192.168.x.x:5060
Looking for *98 in from-sip-internal (domain 192.168.x.x)

<— Transmitting (no NAT) to 192.168.x.x:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.x.x;branch=z9hG4bK4d6318fcc41af6f29;received=192.168.x.x
From: sip:seth@192.168.x.x:5060;tag=fd7db2268d
To: sip:*98@192.168.x.x:5060;tag=as5257c9ea
Call-ID: 89aa93b67b79db9d
CSeq: 20347 SUBSCRIBE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘89aa93b67b79db9d’ Method: SUBSCRIBE

<— SIP read from UDP:192.168.x.x:5062 —>
BYE sip:ata@192.168.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK1053224249;rport
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 34 BYE
Contact: sip:ata@192.168.x.x:5062
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.1B 1.0.5.5
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.x.x:5062 (no NAT)
Scheduling destruction of SIP dialog ‘478414669-5062-4@BJC.BGI.B.BEC’ in 6400 ms (Method: BYE)

<— Transmitting (no NAT) to 192.168.x.x:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.x.x:5062;branch=z9hG4bK1053224249;received=192.168.x.x;rport=5062
From: “SAN FRANCSCO CA” sip:14154499998@192.168.x.x;tag=1156841976
To: sip:ata@192.168.x.x:5060;tag=as63b9e991
Call-ID: 478414669-5062-4@BJC.BGI.B.BEC
CSeq: 34 BYE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (from-trunk, ata, 4) exited non-zero on ‘SIP/ata-00000008’
– Executing [h@from-trunk:1] Hangup(“SIP/ata-00000008”, “”) in new stack
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/ata-00000008’
Really destroying SIP dialog ‘478414669-5062-4@BJC.BGI.B.BEC’ Method: BYE[/code]

I will soon be switching to a VOIP provider and then the fax will come on a specific channel – I will then have to figure that out, but this fax thing from our PSTN is the last thing I need to figure out.

Thanks in advance for any help!

dejanst · March 28, 2013, 3:37pm

What fax software are you using?

Free Fax for Asterisk is really well described here.

sshoemaker · March 28, 2013, 5:34pm

I’m using a Canon MFP Fax connected to an HT503 on the FXS port.

I guess what I don’t understand is how I detect that there is a fax coming in and instead of playing the greeting, it grabs it and sends it to my fax machine (should be a simple exten => probably)…

dejanst · March 29, 2013, 5:05pm

I think that the Fax software you are using should be able to do that. What do the guys from the Canon MFP Fax say?

In any way, I would recommend using a totally separate public telephone number for accepting faxes. Mixing calls and faxes is never a good thing …

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