Re: [asterisk-users] Faxdetect + T38gateway

Jean-Denis Girard Sun, 17 Feb 2013 18:36:50 -0800

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Le 17/02/2013 13:17, Larry Moore a ￩crit :
> You have not provided any information relating to your configurations.


Thanks for your reply. I've tried your settings, no luck.

I'll try to better describe my problem.

T.38 from HT503 to Asterisk + iaxmodem + Hylafax works fine when calling
directly the fax extension (UDPTL traffic is displayed on the Asterisk
console).

faxdetect also works fine, call is redirected to the fax extension (same
as above), and fax is indeed received by Asterisk + iaxmodem + Hylafax.

But in the second case T.38 is not used (no UDPTL trafic on the Asterisk
console), presumably because the T.38 re-invite from the HT503 was sent
and declined by the phone, so HT503 continues sending fax as voice.

This is the SIP dialog, when ht503 calls the phone (100)

<--- SIP read from UDP:192.168.10.170:5060 --->
INVITE sip:100@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.170:5060;branch=z9hG4bK107821504;rport
From: <sip:ht...@asterisk.sysnux.pf>;tag=874861461
To: <sip:1...@asterisk.sysnux.pf>;tag=as702b1ac8
Call-ID: 1925950287-506...@bjc.bgi.ba.bha
CSeq: 42 INVITE
Contact: <sip:ht503@192.168.10.170:5060>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-502 V1.1B 1.0.9.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 274

v=0
o=ht503 8000 8001 IN IP4 192.168.10.170
s=SIP Call
c=IN IP4 192.168.10.170
t=0 0
m=image 5004 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 192.168.10.170:5060 (no NAT)
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Got T.38 offer in SDP in dialog 1925950287-506...@bjc.bgi.ba.bha
Capabilities: us - (gsm|ulaw|alaw|h263|h264|testlaw), peer -
audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0
(nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
Peer doesn't provide video
  == Redirecting 'SIP/ht503-00000038' to fax extension due to peer T.38
re-INVITE

<--- Transmitting (no NAT) to 192.168.10.170:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.10.170:5060;branch=z9hG4bK107821504;received=192.168.10.170;rport=5060
From: <sip:ht...@asterisk.sysnux.pf>;tag=874861461
To: <sip:1...@asterisk.sysnux.pf>;tag=as702b1ac8
Call-ID: 1925950287-506...@bjc.bgi.ba.bha
CSeq: 42 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:100@192.168.0.10:5060>
Content-Length: 0


<------------>
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Spawn extension (stdexten, fax, 1) exited non-zero on
'SIP/ht503-00000038'
    -- Executing [fax@stdexten:1] NoOp("SIP/ht503-00000038", "R￩ception
FAX") in new stack
    -- Executing [fax@stdexten:2] Dial("SIP/ht503-00000038",
"IAX2/iaxmodem0/100") in new stack
    -- Called IAX2/iaxmodem0/100
    -- Call accepted by 127.0.0.1 (format alaw)
    -- Format for call is (alaw)
    -- IAX2/iaxmodem0-2681 is ringing
    -- IAX2/iaxmodem0-2681 answered SIP/ht503-00000038

<--- Reliably Transmitting (no NAT) to 192.168.10.170:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
192.168.10.170:5060;branch=z9hG4bK107821504;received=192.168.10.170;rport=5060
From: <sip:ht...@asterisk.sysnux.pf>;tag=874861461
To: <sip:1...@asterisk.sysnux.pf>;tag=as702b1ac8
Call-ID: 1925950287-506...@bjc.bgi.ba.bha
CSeq: 42 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

Are you sure T.38 is actually used when call is redirected to fax extension?


Thanks,
- -- 
Jean-Denis Girard

SysNux                  Syst￨mes  Linux  en Polyn￩sie fran￧aise
http://www.sysnux.pf/   T￩l: +689 50 10 40 / GSM: +689 79 75 27
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Re: [asterisk-users] Faxdetect + T38gateway

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