More on T.38 fax and Asterisk

Asterisk Asterisk Support
1

Hi everyone

I have posted several items here and on the mailing list regarding T.38 fax. The main concensus is that Asterisk doesn’t do T.38. I think I am not expressing my point very well and/or others are missing my point. This is not a criticism, but I dearly want to get this working and I think if I can explain the problem properly someone out there will have a solution.

OK, Asterisk doesn’t do fax very well. This is true. spandsp is there to receive faxes this is also true. However, this is all good if there are trunk cards in the Asterisk box. I think the assumption here is that every Asterisk system has trunk cards installed internally, most likely from Digium.

However, I use Asterisk purely as as SIP call manager. No cards. I purchase g.729 licences from Digium. Sorry guys, that is all I need from you at present. All my interface to the PSTN is done with SIP gateways. Analog devices like cordless phones and fax machines are connected to small SIP gateways (ATAs). These devices have built in support for fax and it is a standard called T.38 and even though T.38 is not really a codec, it behaves like one.

When I send a fax through my ATA to the PSTN via my ISDN SIP gateway, the gateway that hears the answering fax machine sync tone detects this as fax and converts the audio to data inside the gateway then sends it to the next gateway which re-creates the fax tone at the other end. All Asterisk has to do is ALLOW this to happen. All my gateways have T.38 and they all work fine, but not when registered to Asterisk. I am a relative novice at SIP and I am not able to detect where exactly Asterisk is stopping this T.38 setup. What I do know is that when a SIP gateway detects fax tone, it appears to try to re-negotiate codecs. It trys to replace the G.729, or G.711 or whatever with T.38. Asterisk does not need to be in the signal path, but I think the SIP messages that flow back via Asterisk to renegotiate the codec are met with a forbidden type reply.

I think this is because Asterisk does not know of the rtp payload type of t.38. If this is the case, it should be easy for the clever Asterisk programmers to add in this payload so it can be added to the allow list in sip.conf, etc.

I hope this gives someone a better idea of what I want and maybe we can make it fly.

Cheers

Mark

2

OK, for those who are interested. Here is a sip dump from Asterisk. I have called a fax machine from the PSTN. THe call hits my ISDN SIP gateway calls across to my ATA with a fax machine plugged in. You will see the T.38 protocols come up in the sip info, but I have trouble with this info, so I hope someone can tell me why Asterisk doesn’t want to allow the protocol.

dev*CLI>

Sip read:
REGISTER sip:asterisk:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.127:5060
Contact: sip:250@192.168.44.127:5060;expires=3600
User-Agent: FXS_GW (2asipfxs.106)
From: sip:250@asterisk ;tag=c0a82c7f-13c4-425e4ac0-6b74c-62fb
To: sip:250@asterisk
Call-ID: c0a82c7f-13c4-f-3b83-2ff8-1
CSeq: 22 REGISTER
Content-Length:0

9 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.127 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:250@asterisk ;tag=c0a82c7f-13c4-425e4ac0-6b74c-62fb
To: sip:250@asterisk;tag=as0770b875
Call-ID: c0a82c7f-13c4-f-3b83-2ff8-1
CSeq: 22 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:250@192.168.44.19
Content-Length: 0

to 192.168.44.127:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:250@asterisk ;tag=c0a82c7f-13c4-425e4ac0-6b74c-62fb
To: sip:250@asterisk;tag=as0770b875
Call-ID: c0a82c7f-13c4-f-3b83-2ff8-1
CSeq: 22 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:250@192.168.44.19
WWW-Authenticate: Digest realm=“asterisk”, nonce="10a84417"
Content-Length: 0

to 192.168.44.127:5060
Scheduling destruction of call ‘c0a82c7f-13c4-f-3b83-2ff8-1’ in 15000 ms
dev*CLI>

Sip read:
REGISTER sip:asterisk:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.127:5060
Contact: sip:fax@192.168.44.127:5060;expires=3600
User-Agent: FXS_GW (2asipfxs.106)
From: sip:fax@asterisk ;tag=c0a82c7f-13c4-425e4ac0-6b783-33d7
To: sip:fax@asterisk
Call-ID: c0a82c7f-13c4-f-3b83-2ff8-2
CSeq: 23 REGISTER
Content-Length:0

9 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.127 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:fax@asterisk ;tag=c0a82c7f-13c4-425e4ac0-6b783-33d7
To: sip:fax@asterisk;tag=as77fc8a1c
Call-ID: c0a82c7f-13c4-f-3b83-2ff8-2
CSeq: 23 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:fax@192.168.44.19
Content-Length: 0

to 192.168.44.127:5060
Scheduling destruction of call ‘c0a82c7f-13c4-f-3b83-2ff8-2’ in 15000 ms
dev*CLI>

Sip read:
REGISTER sip:asterisk:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.127:5060
Contact: sip:250@192.168.44.127:5060;expires=3600
Authorization: Digest username=“250”, realm=“asterisk”, nonce=“10a84417”, uri=“sip:asterisk:5060”, response=“2d2e7dd87cf53f186ce12c9ea9ac662e”, algorithm=MD5
User-Agent: FXS_GW (2asipfxs.106)
From: sip:250@asterisk ;tag=c0a82c7f-13c4-425e4ac0-6b7f1-32e2
To: sip:250@asterisk
Call-ID: c0a82c7f-13c4-f-3b83-2ff8-1
CSeq: 24 REGISTER
Content-Length:0

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.127 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:250@asterisk ;tag=c0a82c7f-13c4-425e4ac0-6b7f1-32e2
To: sip:250@asterisk;tag=as0770b875
Call-ID: c0a82c7f-13c4-f-3b83-2ff8-1
CSeq: 24 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:250@192.168.44.19
Content-Length: 0

to 192.168.44.127:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:250@asterisk ;tag=c0a82c7f-13c4-425e4ac0-6b7f1-32e2
To: sip:250@asterisk;tag=as0770b875
Call-ID: c0a82c7f-13c4-f-3b83-2ff8-1
CSeq: 24 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: sip:250@192.168.44.127:5060;expires=3600
Date: Thu, 14 Apr 2005 10:45:20 GMT
Content-Length: 0

to 192.168.44.127:5060
Scheduling destruction of call ‘c0a82c7f-13c4-f-3b83-2ff8-1’ in 15000 ms
dev*CLI>

Sip read:
INVITE sip:666@192.168.44.19 SIP/2.0
CSeq: 1 INVITE
Call-ID: call-004D4E97-1EAB-D911-0200-8@192.168.44.23
Contact: sip:094747310@192.168.44.23
Content-Type: application/sdp
From: sip:094747310@192.168.44.23;tag=c0a82c17-b
Session-GUID: 875705701-825320242-942814565-812016366
To: sip:666@192.168.44.19
Via: SIP/2.0/UDP 192.168.44.23;branch=z9hG4bK-tenor-c0a8-2c17-0014
Content-Length: 227
User-Agent: Quintum/1.0.0
Quintum: 0c01030b023232050100074f1b0001000000112c00040a8080809000000000008300180480818001006c0d80818081303934373437333130007005808136363604038090a31801896c0b01813039343734373331307004813636361301000f0b413031332d303230303330
Max-Forwards: 70

v=0
o=Quintum 12 8 IN IP4 192.168.44.23
s=VoipCall
c=IN IP4 192.168.44.23
t=0 0
m=audio 10256 RTP/AVP 18 8 101
c=IN IP4 192.168.44.23
a=rtpmap:18 g729/8000/1
a=rtpmap:8 pcma/8000/1
a=rtpmap:101 telephone-event/8000/1

13 headers, 10 lines
Using latest request as basis request
Sending to 192.168.44.23 : 5060 (non-NAT)
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.44.23:10256
Found description format g729
Found description format pcma
Found description format telephone-event
Capabilities: us - 0xf07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Found peer '299’
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.44.23;branch=z9hG4bK-tenor-c0a8-2c17-0014
From: sip:094747310@192.168.44.23;tag=c0a82c17-b
To: sip:666@192.168.44.19;tag=as6b9f59e2
Call-ID: call-004D4E97-1EAB-D911-0200-8@192.168.44.23
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:666@192.168.44.19
Proxy-Authenticate: Digest realm=“asterisk”, nonce="35ed8e0b"
Content-Length: 0

to 192.168.44.23:5060
Scheduling destruction of call ‘call-004D4E97-1EAB-D911-0200-8@192.168.44.23’ in 15000 ms
dev*CLI>

Sip read:
ACK sip:666@192.168.44.19 SIP/2.0
CSeq: 1 ACK
Call-ID: call-004D4E97-1EAB-D911-0200-8@192.168.44.23
Contact: sip:094747310@192.168.44.23
From: sip:094747310@192.168.44.23;tag=c0a82c17-b
Session-GUID: 875705701-825320242-942814565-812016366
To: sip:666@192.168.44.19;tag=as6b9f59e2
Via: SIP/2.0/UDP 192.168.44.23;branch=z9hG4bK-tenor-c0a8-2c17-0014
User-Agent: Quintum/1.0.0
Quintum: 0c01030b023232050100074f1b0001000000112c00040a8080809000000000008300180480818001006c0d80818081303934373437333130007005808136363604038090a31801896c0b01813039343734373331307004813636361301000f0b413031332d303230303330
Max-Forwards: 70

11 headers, 0 lines
dev*CLI>

Sip read:
INVITE sip:666@192.168.44.19 SIP/2.0
CSeq: 2 INVITE
Call-ID: call-004D4E97-1EAB-D911-0200-8@192.168.44.23
Contact: sip:094747310@192.168.44.23
Content-Type: application/sdp
From: sip:094747310@192.168.44.23;tag=c0a82c17-b
Proxy-Authorization: Digest realm=“asterisk”, nonce=“35ed8e0b”, username=“299”, uri="sip:666@192.168.44.19", response="df1e9fca1de80a3960d6f24d2e747f14"
Session-GUID: 875705701-825320242-942814565-812016366
To: sip:666@192.168.44.19
Via: SIP/2.0/UDP 192.168.44.23;branch=z9hG4bK-tenor-c0a8-2c17-0015
Content-Length: 227
User-Agent: Quintum/1.0.0
Quintum: 0c01030b023232050100074f1b0001000000112c00040a8080809000000000008300180480818001006c0d80818081303934373437333130007005808136363604038090a31801896c0b01813039343734373331307004813636361301000f0b413031332d303230303330
Max-Forwards: 70

v=0
o=Quintum 12 8 IN IP4 192.168.44.23
s=VoipCall
c=IN IP4 192.168.44.23
t=0 0
m=audio 10256 RTP/AVP 18 8 101
c=IN IP4 192.168.44.23
a=rtpmap:18 g729/8000/1
a=rtpmap:8 pcma/8000/1
a=rtpmap:101 telephone-event/8000/1

14 headers, 10 lines
Using latest request as basis request
Sending to 192.168.44.23 : 5060 (non-NAT)
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.44.23:10256
Found description format g729
Found description format pcma
Found description format telephone-event
Capabilities: us - 0xf07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Found peer '299’
Looking for 666 in from-pstn
list_route: hop: sip:094747310@192.168.44.23
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.44.23;branch=z9hG4bK-tenor-c0a8-2c17-0015
From: sip:094747310@192.168.44.23;tag=c0a82c17-b
To: sip:666@192.168.44.19;tag=as04455d24
Call-ID: call-004D4E97-1EAB-D911-0200-8@192.168.44.23
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:666@192.168.44.19
Content-Length: 0

to 192.168.44.23:5060
We’re at 192.168.44.19 port 13890
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 9 lines
Reliably Transmitting:
INVITE sip:250@192.168.44.127:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.19:5060;branch=z9hG4bK7b24953c
From: “094747310” sip:094747310@192.168.44.19;tag=as58dd6a1b
To: sip:250@192.168.44.127:5060
Contact: sip:094747310@192.168.44.19
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 14 Apr 2005 10:45:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 192

v=0
o=root 3165 3165 IN IP4 192.168.44.19
s=session
c=IN IP4 192.168.44.19
t=0 0
m=audio 13890 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 192.168.44.127:5060
dev*CLI>

Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.44.19:5060;branch=z9hG4bK7b24953c
Contact: sip:250@192.168.44.127:5060
User-Agent: FXS_GW (2asipfxs.106)
From: "094747310"sip:094747310@192.168.44.19 ;tag=as58dd6a1b
To: sip:250@192.168.44.127:5060 ;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 102 INVITE
Content-Length:0

9 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.44.23;branch=z9hG4bK-tenor-c0a8-2c17-0015
From: sip:094747310@192.168.44.23;tag=c0a82c17-b
To: sip:666@192.168.44.19;tag=as04455d24
Call-ID: call-004D4E97-1EAB-D911-0200-8@192.168.44.23
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:666@192.168.44.19
Content-Length: 0

to 192.168.44.23:5060
dev*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.19:5060;branch=z9hG4bK7b24953c
Contact: sip:250@192.168.44.127:5060
User-Agent: FXS_GW (2asipfxs.106)
From: "094747310"sip:094747310@192.168.44.19 ;tag=as58dd6a1b
To: sip:250@192.168.44.127:5060 ;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length:134

v=0
o=FXS_GW 12367 0 IN IP4 192.168.44.127
s=Audio Session
i=Audio Session
c=IN IP4 192.168.44.127
t=0 0
m=audio 16384 RTP/AVP

10 headers, 7 lines
Peer audio RTP is at port 192.168.44.127:16384
Capabilities: us - 0x0 (nothing), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: sip:250@192.168.44.127:5060
set_destination: Parsing sip:250@192.168.44.127:5060 for address/port to send to
set_destination: set destination to 192.168.44.127, port 5060
Transmitting:
ACK sip:250@192.168.44.127:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.19:5060;branch=z9hG4bK43381ad8
From: “094747310” sip:094747310@192.168.44.19;tag=as58dd6a1b
To: sip:250@192.168.44.127:5060;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
Contact: sip:094747310@192.168.44.19
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 192.168.44.127:5060
We’re at 192.168.44.19 port 13282
Answering with preferred capability 0x100 (g729)
Answering with preferred capability 0x400 (ilbc)
Answering with preferred capability 0x8 (alaw)
Answering with capability 0x1 (g723)
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x10 (g726)
Answering with capability 0x20 (adpcm)
Answering with capability 0x40 (slin)
Answering with capability 0x80 (lpc10)
Answering with capability 0x200 (speex)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.23;branch=z9hG4bK-tenor-c0a8-2c17-0015
From: sip:094747310@192.168.44.23;tag=c0a82c17-b
To: sip:666@192.168.44.19;tag=as04455d24
Call-ID: call-004D4E97-1EAB-D911-0200-8@192.168.44.23
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:666@192.168.44.19
Content-Type: application/sdp
Content-Length: 467

v=0
o=root 3165 3165 IN IP4 192.168.44.19
s=session
c=IN IP4 192.168.44.19
t=0 0
m=audio 13282 RTP/AVP 18 97 8 4 3 0 2 5 10 7 110 101
a=rtpmap:18 G729/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

to 192.168.44.23:5060
dev*CLI>

Sip read:
ACK sip:666@192.168.44.19 SIP/2.0
CSeq: 2 ACK
Call-ID: call-004D4E97-1EAB-D911-0200-8@192.168.44.23
Contact: sip:094747310@192.168.44.23
From: sip:094747310@192.168.44.23;tag=c0a82c17-b
Proxy-Authorization: Digest realm=“asterisk”, nonce=“35ed8e0b”, username=“299”, uri="sip:666@192.168.44.19", response="df1e9fca1de80a3960d6f24d2e747f14"
Session-GUID: 875705701-825320242-942814565-812016366
To: sip:666@192.168.44.19;tag=as04455d24
Via: SIP/2.0/UDP 192.168.44.23;branch=z9hG4bK-tenor-c0a8-2c17-0015
User-Agent: Quintum/1.0.0
Quintum: 0c01030b023232050100074f1b0001000000112c00040a8080809000000000008300180480818001006c0d80818081303934373437333130007005808136363604038090a31801896c0b01813039343734373331307004813636361301000f0b413031332d303230303330
Max-Forwards: 70

12 headers, 0 lines
dev*CLI>

Sip read:
INVITE sip:094747310@192.168.44.19 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.127:5060
Contact: sip:250@192.168.44.127:5060
User-Agent: FXS_GW (2asipfxs.106)
From: sip:250@192.168.44.127:5060 ;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
To: "094747310"sip:094747310@192.168.44.19 ;tag=as58dd6a1b
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length:375

v=0
o=FXS_GW 12367 0 IN IP4 192.168.44.127
s=Audio Session
i=Audio Session
c=IN IP4 192.168.44.127
t=0 0
m=image 16384 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:72
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy

10 headers, 16 lines
Using latest request as basis request
Sending to 192.168.44.127 : 5060 (non-NAT)
Capabilities: us - 0x0 (nothing), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
dev*CLI>

Sip read:
INVITE sip:094747310@192.168.44.19 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.127:5060
Contact: sip:250@192.168.44.127:5060
User-Agent: FXS_GW (2asipfxs.106)
From: sip:250@192.168.44.127:5060 ;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
To: "094747310"sip:094747310@192.168.44.19 ;tag=as58dd6a1b
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length:375

v=0
o=FXS_GW 12367 0 IN IP4 192.168.44.127
s=Audio Session
i=Audio Session
c=IN IP4 192.168.44.127
t=0 0
m=image 16384 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:72
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy

10 headers, 16 lines
Ignoring this request
We’re at 192.168.44.19 port 13890
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:250@192.168.44.127:5060 ;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
To: "094747310"sip:094747310@192.168.44.19 ;tag=as58dd6a1b
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:094747310@192.168.44.19
Content-Type: application/sdp
Content-Length: 136

v=0
o=root 3165 3166 IN IP4 192.168.44.19
s=session
c=IN IP4 192.168.44.19
t=0 0
m=audio 13890 RTP/AVP
a=silenceSupp:off - - - -

to 192.168.44.127:5060
Destroying call 'c0a82c7f-13c4-f-3b83-2ff8-2’
Destroying call 'c0a82c7f-13c4-f-3b83-2ff8-1’
Retransmitting #1 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:250@192.168.44.127:5060 ;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
To: "094747310"sip:094747310@192.168.44.19 ;tag=as58dd6a1b
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:094747310@192.168.44.19
Content-Type: application/sdp
Content-Length: 136

v=0
o=root 3165 3166 IN IP4 192.168.44.19
s=session
c=IN IP4 192.168.44.19
t=0 0
m=audio 13890 RTP/AVP
a=silenceSupp:off - - - -

to 192.168.44.127:5060
Retransmitting #2 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:250@192.168.44.127:5060 ;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
To: "094747310"sip:094747310@192.168.44.19 ;tag=as58dd6a1b
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:094747310@192.168.44.19
Content-Type: application/sdp
Content-Length: 136

v=0
o=root 3165 3166 IN IP4 192.168.44.19
s=session
c=IN IP4 192.168.44.19
t=0 0
m=audio 13890 RTP/AVP
a=silenceSupp:off - - - -

to 192.168.44.127:5060
Retransmitting #3 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:250@192.168.44.127:5060 ;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
To: "094747310"sip:094747310@192.168.44.19 ;tag=as58dd6a1b
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:094747310@192.168.44.19
Content-Type: application/sdp
Content-Length: 136

v=0
o=root 3165 3166 IN IP4 192.168.44.19
s=session
c=IN IP4 192.168.44.19
t=0 0
m=audio 13890 RTP/AVP
a=silenceSupp:off - - - -

to 192.168.44.127:5060
Retransmitting #4 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:250@192.168.44.127:5060 ;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
To: "094747310"sip:094747310@192.168.44.19 ;tag=as58dd6a1b
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:094747310@192.168.44.19
Content-Type: application/sdp
Content-Length: 136

v=0
o=root 3165 3166 IN IP4 192.168.44.19
s=session
c=IN IP4 192.168.44.19
t=0 0
m=audio 13890 RTP/AVP
a=silenceSupp:off - - - -

to 192.168.44.127:5060
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.127:5060
From: sip:250@192.168.44.127:5060 ;tag=c0a82c7f-13c4-425e4ac2-6bea4-1b2e
To: "094747310"sip:094747310@192.168.44.19 ;tag=as58dd6a1b
Call-ID: 1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:094747310@192.168.44.19
Content-Type: application/sdp
Content-Length: 136

v=0
o=root 3165 3166 IN IP4 192.168.44.19
s=session
c=IN IP4 192.168.44.19
t=0 0
m=audio 13890 RTP/AVP
a=silenceSupp:off - - - -

to 192.168.44.127:5060
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:line1@asterisk;tag=0-13c4-425e4ad1-60bfd19-74f5
To: sip:line1@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11159 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad1-60bfd19-228a
Contact: sip:line1@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.8 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad1-60bfd19-228a
From: sip:line1@asterisk;tag=0-13c4-425e4ad1-60bfd19-74f5
To: sip:line1@asterisk;tag=as62bbbf91
Call-ID: 0-13c4-0-b31-210e
CSeq: 11159 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:line1@192.168.44.19
Content-Length: 0

to 192.168.44.8:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad1-60bfd19-228a
From: sip:line1@asterisk;tag=0-13c4-425e4ad1-60bfd19-74f5
To: sip:line1@asterisk;tag=as62bbbf91
Call-ID: 0-13c4-0-b31-210e
CSeq: 11159 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:line1@192.168.44.19
WWW-Authenticate: Digest realm=“asterisk”, nonce="7eef2ab2"
Content-Length: 0

to 192.168.44.8:5060
Scheduling destruction of call ‘0-13c4-0-b31-210e’ in 15000 ms
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:line2@asterisk;tag=0-13c4-425e4ad1-60bff4e-6b71
To: sip:line2@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11160 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad1-60bff4e-5d56
Contact: sip:line2@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.8 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad1-60bff4e-5d56
From: sip:line2@asterisk;tag=0-13c4-425e4ad1-60bff4e-6b71
To: sip:line2@asterisk;tag=as62bbbf91
Call-ID: 0-13c4-0-b31-210e
CSeq: 11160 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:line2@192.168.44.19
Content-Length: 0

to 192.168.44.8:5060
Scheduling destruction of call ‘0-13c4-0-b31-210e’ in 15000 ms
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:nil@asterisk;tag=0-13c4-425e4ad2-60c0183-4dad
To: sip:nil@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11161 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad2-60c0183-78e2
Contact: sip:nil@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.8 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad2-60c0183-78e2
From: sip:nil@asterisk;tag=0-13c4-425e4ad2-60c0183-4dad
To: sip:nil@asterisk;tag=as62bbbf91
Call-ID: 0-13c4-0-b31-210e
CSeq: 11161 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:nil@192.168.44.19
Content-Length: 0

to 192.168.44.8:5060
Scheduling destruction of call ‘0-13c4-0-b31-210e’ in 15000 ms
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:nil@asterisk;tag=0-13c4-425e4ad2-60c03b8-17a9
To: sip:nil@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11162 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad2-60c03b8-412e
Contact: sip:nil@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.8 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad2-60c03b8-412e
From: sip:nil@asterisk;tag=0-13c4-425e4ad2-60c03b8-17a9
To: sip:nil@asterisk;tag=as62bbbf91
Call-ID: 0-13c4-0-b31-210e
CSeq: 11162 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:nil@192.168.44.19
Content-Length: 0

to 192.168.44.8:5060
Scheduling destruction of call ‘0-13c4-0-b31-210e’ in 15000 ms
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:nil@asterisk;tag=0-13c4-425e4ad3-60c05ed-565
To: sip:nil@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11163 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad3-60c05ed-423a
Contact: sip:nil@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.8 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad3-60c05ed-423a
From: sip:nil@asterisk;tag=0-13c4-425e4ad3-60c05ed-565
To: sip:nil@asterisk;tag=as62bbbf91
Call-ID: 0-13c4-0-b31-210e
CSeq: 11163 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:nil@192.168.44.19
Content-Length: 0

to 192.168.44.8:5060
Scheduling destruction of call ‘0-13c4-0-b31-210e’ in 15000 ms
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:nil@asterisk;tag=0-13c4-425e4ad3-60c0822-12e1
To: sip:nil@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11164 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad3-60c0822-4806
Contact: sip:nil@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.8 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad3-60c0822-4806
From: sip:nil@asterisk;tag=0-13c4-425e4ad3-60c0822-12e1
To: sip:nil@asterisk;tag=as62bbbf91
Call-ID: 0-13c4-0-b31-210e
CSeq: 11164 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:nil@192.168.44.19
Content-Length: 0

to 192.168.44.8:5060
Scheduling destruction of call ‘0-13c4-0-b31-210e’ in 15000 ms
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:line1@asterisk;tag=0-13c4-425e4ad1-60bfd19-74f5
To: sip:line1@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11159 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad1-60bfd19-228a
Contact: sip:line1@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:line2@asterisk;tag=0-13c4-425e4ad1-60bff4e-6b71
To: sip:line2@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11160 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad1-60bff4e-5d56
Contact: sip:line2@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:nil@asterisk;tag=0-13c4-425e4ad2-60c0183-4dad
To: sip:nil@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11161 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad2-60c0183-78e2
Contact: sip:nil@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:nil@asterisk;tag=0-13c4-425e4ad2-60c03b8-17a9
To: sip:nil@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11162 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad2-60c03b8-412e
Contact: sip:nil@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:nil@asterisk;tag=0-13c4-425e4ad3-60c05ed-565
To: sip:nil@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11163 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad3-60c05ed-423a
Contact: sip:nil@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:nil@asterisk;tag=0-13c4-425e4ad3-60c0822-12e1
To: sip:nil@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11164 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad3-60c0822-4806
Contact: sip:nil@192.168.44.8:5060;q=0.5
Expires: 3600
User-Agent: FXO_GW
Content-Length:0

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.8 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad3-60c0822-4806
From: sip:nil@asterisk;tag=0-13c4-425e4ad3-60c0822-12e1
To: sip:nil@asterisk;tag=as62bbbf91
Call-ID: 0-13c4-0-b31-210e
CSeq: 11164 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:nil@192.168.44.19
Content-Length: 0

to 192.168.44.8:5060
Scheduling destruction of call ‘0-13c4-0-b31-210e’ in 15000 ms
dev*CLI>

Sip read:
REGISTER sip:asterisk SIP/2.0
From: sip:line1@asterisk;tag=0-13c4-425e4ad4-60c0ad4-3cf4
To: sip:line1@asterisk
Call-ID: 0-13c4-0-b31-210e
CSeq: 11165 REGISTER
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad4-60c0ad9-7c1d
Contact: sip:line1@192.168.44.8:5060;q=0.5
Expires: 3600
Authorization: Digest username=“line1”, realm=“asterisk”, nonce=“7eef2ab2”, uri=“sip:asterisk”, response=“c51c84d1c4cbf989ca9d9881dab8ff6e”, algorithm=MD5
User-Agent: FXO_GW
Content-Length:0

dev*CLI>
11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.44.8 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad4-60c0ad9-7c1d
From: sip:line1@asterisk;tag=0-13c4-425e4ad4-60c0ad4-3cf4
To: sip:line1@asterisk;tag=as62bbbf91
Call-ID: 0-13c4-0-b31-210e
CSeq: 11165 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:line1@192.168.44.19
Content-Length: 0

to 192.168.44.8:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.8:5060;branch=z9hG4bK-425e4ad4-60c0ad9-7c1d
From: sip:line1@asterisk;tag=0-13c4-425e4ad4-60c0ad4-3cf4
To: sip:line1@asterisk;tag=as62bbbf91
Call-ID: 0-13c4-0-b31-210e
CSeq: 11165 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: sip:line1@192.168.44.8:5060;expires=3600
Date: Thu, 14 Apr 2005 10:45:44 GMT
Content-Length: 0

to 192.168.44.8:5060
Scheduling destruction of call ‘0-13c4-0-b31-210e’ in 15000 ms
dev*CLI>

Sip read:
BYE sip:666@192.168.44.19 SIP/2.0
CSeq: 3 BYE
Call-ID: call-004D4E97-1EAB-D911-0200-8@192.168.44.23
From: sip:094747310@192.168.44.23;tag=c0a82c17-b
To: sip:666@192.168.44.19;tag=as04455d24
Via: SIP/2.0/UDP 192.168.44.23;branch=z9hG4bK-tenor-c0a8-2c17-0016
User-Agent: Quintum/1.0.0
Max-Forwards: 70

8 headers, 0 lines
Sending to 192.168.44.23 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.44.23;branch=z9hG4bK-tenor-c0a8-2c17-0016
From: sip:094747310@192.168.44.23;tag=c0a82c17-b
To: sip:666@192.168.44.19;tag=as04455d24
Call-ID: call-004D4E97-1EAB-D911-0200-8@192.168.44.23
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:666@192.168.44.19
Content-Length: 0

to 192.168.44.23:5060
Destroying call '1f2219f325bb28a12ae3b51d540c4a7f@192.168.44.19’
Destroying call 'call-004D4E97-1EAB-D911-0200-8@192.168.44.23’
dev*CLI>

3

I too would love to see this implemented, T.38 passthrough so I can use my Quintum Tenor A800 as I had orginally hoped.