[asterisk-users] Problem T.38

Sat May 30 16:38:07 CDT 2009

Boa Tarde Lista.

  I'm having problems in tramissão a fax using T.38.

   My scenario is:
   Asterisk 1.6.0.5
   2 ATA of Intelbras 2210.
   ReceiveFAX in the asterisk.

   Unable to fax when it is a ATA to another user on the Asterisk means, if
I directly between the ATA works perfectly, is a step to the ATA ReceiveFAX
of Asterisk works perfect, but if I try to pass between two Branches using
the same ATA does not work ever.

 rtp.conf:
[general]
rtpstart=17000
rtpend=33000

udptl.conf:
[general]
udptlstart=4000
udptlend=4999
T38FaxUdpEC = t38UDPRedundancy
T38FaxMaxDatagram = 400
udptlfecentries = 3
udptlfecspan = 3

sip.conf:
t38pt_udptl = yes
t38pt_rtp=no
t38pt_tcp=no

I did a tcpdump and when after listening to the signal for a fax
transmission, the network traffic for RTP, and only the end of the
connection timeout in the show some more data from SIP protocol and the
connection drops.

Example of the signal is given when a fax:
18:05:28.931701 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 439
18:05:29.047186 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 573
18:05:29.163231 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 439
18:05:31.336965 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port
17003 unreachable, length 36
18:05:36.339933 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port
17003 unreachable, length 36
18:05:41.338790 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port
17003 unreachable, length 36

But when using ATA ReceiveFAX for the RTP traffic is constant until the
passage of the fax.

The next sip debug peer at the time that the fax is not:

<--- Transmitting (NAT) to XX.XX.XX.67:1024 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.100:5060
;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67
From: 2005618<sip:2005618 at XX.XX.XX.66>;tag=74tf2-7Q4xE0
To: sip:2007 at XX.XX.XX.66;tag=as46031e07
Call-ID: CcD6S0-VFf0Z8f2 at XX.XX.XX.66
CSeq: 51 INVITE
User-Agent: Asterisk PBX 1.6.0.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:2007 at XX.XX.XX.66>
Content-Length: 0

<------------>
Audio is at XX.XX.XX.66 port 30206
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
engeplus*CLI>
<--- Transmitting (NAT) to XX.XX.XX.67:1024 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.100:5060
;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67
From: 2005618<sip:2005618 at XX.XX.XX.66>;tag=74tf2-7Q4xE0
To: sip:2007 at XX.XX.XX.66;tag=as46031e07
Call-ID: CcD6S0-VFf0Z8f2 at XX.XX.XX.66
CSeq: 51 INVITE
User-Agent: Asterisk PBX 1.6.0.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:2007 at XX.XX.XX.66>
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 577489170 577489170 IN IP4 XX.XX.XX.66
s=Asterisk PBX 1.6.0.5
c=IN IP4 XX.XX.XX.66
t=0 0
m=audio 30206 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- SIP/2007618-b7cb7cf8 is ringing
    -- SIP/2007618-b7cb7cf8 answered SIP/2005618-08bd3e50
Audio is at XX.XX.XX.66 port 30206
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to XX.XX.XX.67:1024 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.100:5060
;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67
From: 2005618<sip:2005618 at XX.XX.XX.66>;tag=74tf2-7Q4xE0
To: sip:2007 at XX.XX.XX.66;tag=as46031e07
Call-ID: CcD6S0-VFf0Z8f2 at XX.XX.XX.66
CSeq: 51 INVITE
User-Agent: Asterisk PBX 1.6.0.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:2007 at XX.XX.XX.66>
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 577489170 577489171 IN IP4 XX.XX.XX.66
s=Asterisk PBX 1.6.0.5
c=IN IP4 XX.XX.XX.66
t=0 0
m=audio 30206 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
engeplus*CLI>
<--- SIP read from UDP://XX.XX.XX.67:1024 --->
ACK sip:2007 at XX.XX.XX.66:5060 SIP/2.0
From: 2005618<sip:2005618 at XX.XX.XX.66>;tag=74tf2-7Q4xE0
To: sip:2007 at XX.XX.XX.66;tag=as46031e07
Call-ID: CcD6S0-VFf0Z8f2 at XX.XX.XX.66
CSeq: 51 ACK
Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bKdN0f2-iNjO1len
Contact: 2005618<sip:2005618 at 192.168.2.100:5060>
Max-Forwards: 70
User-Agent: INTELBRAS ATA GKM2210T -  Nov 19 2008
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:2005618 at 192.168.2.100:5060> for address/port
to send to
set_destination: set destination to 192.168.2.100, port 5060
Reliably Transmitting (NAT) to XX.XX.XX.67:1024:
INVITE sip:2005618 at 192.168.2.100:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc;rport
Max-Forwards: 70
From: sip:2007 at XX.XX.XX.66;tag=as46031e07
To: 2005618<sip:2005618 at XX.XX.XX.66>;tag=74tf2-7Q4xE0
Contact: <sip:2007 at XX.XX.XX.66>
Call-ID: CcD6S0-VFf0Z8f2 at XX.XX.XX.66
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 577489170 577489172 IN IP4 XX.XX.XX.66
s=Asterisk PBX 1.6.0.5
c=IN IP4 XX.XX.XX.66
t=0 0
m=image 4729 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
engeplus*CLI>
<--- SIP read from UDP://XX.XX.XX.67:1024 --->
SIP/2.0 100 Trying
From: sip:2007 at XX.XX.XX.66;tag=as46031e07
To: 2005618<sip:2005618 at XX.XX.XX.66>;tag=74tf2-7Q4xE0
Call-ID: CcD6S0-VFf0Z8f2 at XX.XX.XX.66
CSeq: 102 INVITE
Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc
Content-Length: 0

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

<--- SIP read from UDP://XX.XX.XX.67:1024 --->
SIP/2.0 200 OK
From: sip:2007 at XX.XX.XX.66;tag=as46031e07
To: 2005618<sip:2005618 at XX.XX.XX.66>;tag=74tf2-7Q4xE0
Call-ID: CcD6S0-VFf0Z8f2 at XX.XX.XX.66
CSeq: 102 INVITE
Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc
Contact: 2005618<sip:2005618 at 192.168.2.100:5060>
User-Agent: INTELBRAS ATA GKM2210T -  Nov 19 2008
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER
Supported: timer,replaces
Content-Type: application/sdp
Content-Length: 243

v=0
o=2005618 207176 2 IN IP4 192.168.2.100
s=-
c=IN IP4 192.168.2.100
t=0 0
m=image 17002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:400
a=T38FaxUdpEc:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF

<------------->
--- (12 headers 11 lines) ---
Got T.38 offer in SDP in dialog CcD6S0-VFf0Z8f2 at XX.XX.XX.66
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
Callid CcD6S0-VFf0Z8f2 at XX.XX.XX.66
Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:2005618 at 192.168.2.100:5060> for address/port
to send to
set_destination: set destination to 192.168.2.100, port 5060
Transmitting (NAT) to XX.XX.XX.67:1024:
ACK sip:2005618 at 192.168.2.100:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK732ec166;rport
Max-Forwards: 70
From: sip:2007 at XX.XX.XX.66;tag=as46031e07
To: 2005618<sip:2005618 at XX.XX.XX.66>;tag=74tf2-7Q4xE0
Contact: <sip:2007 at XX.XX.XX.66>
Call-ID: CcD6S0-VFf0Z8f2 at XX.XX.XX.66
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.5
Content-Length: 0

---
engeplus*CLI>

How has the idea to do? Have tried several configurations possible.
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