Поставил два Asterisk 1.6.2.2 на соседние машины и пытаюсь организовать между ними обмен факсами по sip. Факсы передаются и принимаются, проблема лишь в том, что обмен идет по аудио каналу, а не по t.38, хотя в разделе [general] файла sip.conf стоит t38pt_udptl=yes
В чем проблема?

Проблема в том, что почему-то не хочется включить sip debug на обмен факсами и посмотреть - что же там передаётся в приложении sdp пакета INVITE на самом деле.

А вот что:
*CLI> originate local/1104@pstnoutgoing application SendFax /home/eliduc/fax2.tiff
-- Executing [1104@pstnoutgoing:1] Dial("Local/1104@pstnoutgoing-363b;2", "SIP/asterisk2/1104,30") in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
Audio is at 172.16.1.44 port 11700
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.16.1.4:5060:
INVITE sip:1104@172.16.1.4 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.44:5060;branch=z9hG4bK46d42e9c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
To: <sip:1104@172.16.1.4>
Contact: <sip:asterisk2@172.16.1.44>
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Date: Tue, 30 Mar 2010 12:12:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1953966858 1953966858 IN IP4 172.16.1.44
s=Asterisk PBX 1.6.2.2
c=IN IP4 172.16.1.44
t=0 0
m=audio 11700 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

---
-- Called asterisk2/1104

<--- SIP read from UDP:172.16.1.4:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.44:5060;branch=z9hG4bK46d42e9c;received=172.16.1.44;rport=5060
From: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
To: <sip:1104@172.16.1.4>
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1104@172.16.1.4>
Content-Length: 0


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

<--- SIP read from UDP:172.16.1.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.44:5060;branch=z9hG4bK46d42e9c;received=172.16.1.44;rport=5060
From: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
To: <sip:1104@172.16.1.4>;tag=as47e01613
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1104@172.16.1.4>
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 345185398 345185398 IN IP4 172.16.1.4
s=Asterisk PBX 1.6.2.2
c=IN IP4 172.16.1.4
t=0 0
m=audio 19498 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

<------------->
--- (14 headers 13 lines) ---
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 - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.1.4:19498
list_route: hop: <sip:1104@172.16.1.4>
set_destination: Parsing <sip:1104@172.16.1.4> for address/port to send to
set_destination: set destination to 172.16.1.4, port 5060
Transmitting (NAT) to 172.16.1.4:5060:
ACK sip:1104@172.16.1.4 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.44:5060;branch=z9hG4bK5f5c0838;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
To: <sip:1104@172.16.1.4>;tag=as47e01613
Contact: <sip:asterisk2@172.16.1.44>
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0


---
-- SIP/asterisk2-00000000 answered Local/1104@pstnoutgoing-363b;2

<--- SIP read from UDP:172.16.1.4:5060 --->
INVITE sip:asterisk2@172.16.1.44 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK22d49e46;rport
Max-Forwards: 70
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Contact: <sip:1104@172.16.1.4>
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 345185398 345185399 IN IP4 172.16.1.4
s=Asterisk PBX 1.6.2.2
c=IN IP4 172.16.1.4
t=0 0
m=image 4671 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxTranscodingMMR
a=T38FaxTranscodingJBIG
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPFEC

<------------->
--- (17 headers 14 lines) ---
Sending to 172.16.1.4 : 5060 (NAT)
Got T.38 offer in SDP in dialog 590071832e13a6c04cae8d441241de03@172.16.1.44
Capabilities: us - 0xc (ulaw|alaw), 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)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

<--- Transmitting (NAT) to 172.16.1.4:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK22d49e46;received=172.16.1.4;rport=5060
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk2@172.16.1.44>
Content-Length: 0


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

<--- Reliably Transmitting (NAT) to 172.16.1.4:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK22d49e46;received=172.16.1.4;rport=5060
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


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

<--- SIP read from UDP:172.16.1.4:5060 --->
ACK sip:asterisk2@172.16.1.44 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK22d49e46;rport
Max-Forwards: 70
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Contact: <sip:1104@172.16.1.4>
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0


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

<--- SIP read from UDP:172.16.1.4:5060 --->
INVITE sip:asterisk2@172.16.1.44 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK1f65faf3;rport
Max-Forwards: 70
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Contact: <sip:1104@172.16.1.4>
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 345185398 345185400 IN IP4 172.16.1.4
s=Asterisk PBX 1.6.2.2
c=IN IP4 172.16.1.4
t=0 0
m=audio 19498 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

<------------->
--- (17 headers 13 lines) ---
Sending to 172.16.1.4 : 5060 (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 - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.1.4:19498

<--- Transmitting (NAT) to 172.16.1.4:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK1f65faf3;received=172.16.1.4;rport=5060
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 103 INVITE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk2@172.16.1.44>
Content-Length: 0


<------------>
Audio is at 172.16.1.44 port 11700
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 172.16.1.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK1f65faf3;received=172.16.1.4;rport=5060
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 103 INVITE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk2@172.16.1.44>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1953966858 1953966859 IN IP4 172.16.1.44
s=Asterisk PBX 1.6.2.2
c=IN IP4 172.16.1.44
t=0 0
m=audio 11700 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:172.16.1.4:5060 --->
ACK sip:asterisk2@172.16.1.44 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK2bfbc8d7;rport
Max-Forwards: 70
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Contact: <sip:1104@172.16.1.4>
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
*CLI> -- Launching SendFax(/home/eliduc/fax2.tiff) on SIP/asterisk2-00000000
[Mar 30 16:12:54] NOTICE[24352]: app_fax.c:795 transmit: FAXMODE=audio
*CLI> == Spawn extension (pstnoutgoing, 1104, 1) exited non-zero on 'Local/1104@pstnoutgoing-363b;2'

<--- SIP read from UDP:172.16.1.4:5060 --->
BYE sip:asterisk2@172.16.1.44 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK0c662fc1;rport
Max-Forwards: 70
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.2
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 172.16.1.4 : 5060 (NAT)

<--- Transmitting (NAT) to 172.16.1.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK0c662fc1;received=172.16.1.4;rport=5060
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 104 BYE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 30 16:12:54] WARNING[24352]: app_fax.c:813 transmit: Transmission error
Really destroying SIP dialog '590071832e13a6c04cae8d441241de03@172.16.1.44' Method: BYE

Теперь проблема в том, что не хочется анализировать лог? Читать RFC 4612 где написано, что t38 передаётся вот так в SDP:

m=audio 6800 RTP/AVP 0 98 a=rtpmap:98 t38/8000 a=fmtp:98
T38FaxVersion=2;T38FaxRateManagement=transferredTCF

Кстати, если Reliably Transmitting (NAT) to 172.16.1.4 то это не соседние машины.

Т.е., как я понимаю, принимающая сторона не хочет принимать по t.38, вот что она пишет:
...
-- Executing [fax@recvfax:3] ReceiveFAX("SIP/asterisk-00000000", "/tmp/1104-asterisk2-0330161602.tif") in new stack
set_destination: Parsing <sip:asterisk2@172.16.1.44> for address/port to send to
set_destination: set destination to 172.16.1.44, port 5060
Reliably Transmitting (NAT) to 172.16.1.44:5060:
INVITE sip:asterisk2@172.16.1.44 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK22d49e46;rport
Max-Forwards: 70
From: <sip:1104@172.16.1.4>;tag=as47e01613
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as05acbdd6
Contact: <sip:1104@172.16.1.4>
Call-ID: 590071832e13a6c04cae8d441241de03@172.16.1.44
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 345185398 345185399 IN IP4 172.16.1.4
s=Asterisk PBX 1.6.2.2
c=IN IP4 172.16.1.4
t=0 0
m=image 4671 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxTranscodingMMR
a=T38FaxTranscodingJBIG
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPFEC

---
....
с чем это может быть связано?

А машины всё-таки соседние, несмотря на nat (а разве nat налагает какие-то дополнительные ограничения на использование t.38?).

Не знаю, зачем Вам НАТ, если машины соседние
asterisk2@172.16.1.44
и
1104@172.16.1.4
Зачем между ними НАТ устраивать то?

Ну а по проблеме: там написано в пакете
SIP re-invite
Откройте для себя что такое T38 re-invite.

Это тестовая конфигурация.

ded:

Ну а по проблеме: там написано в пакете
SIP re-invite
Откройте для себя что такое T38 re-invite.

Спасибо за совет. Завтра я попробую сделать это маленькое открытие.

Я попробовал отправить факс с факс-аппарата (через linksys) и оказалось, что они отлично передаются по t.38, т.е. в процессе передачи * сообщает о готовности принимать по t.38:

set_destination: Parsing <sip:1104@172.16.1.5:5060> for address/port to send to
set_destination: set destination to 172.16.1.5, port 5060
Reliably Transmitting (no NAT) to 172.16.1.5:5060:
INVITE sip:1104@172.16.1.5:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.44:5060;branch=z9hG4bK09d13e17;rport
Max-Forwards: 70
From: <sip:122@172.16.1.44>;tag=as57e290fa
To: 1104 <sip:1104@172.16.1.44>;tag=598ca75348a42654o0
Contact: <sip:122@172.16.1.44>
Call-ID: 6a928e7f-dafd8918@172.16.1.5
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 334

v=0
o=root 833855041 833855042 IN IP4 172.16.1.44
s=Asterisk PBX 1.6.2.2
c=IN IP4 172.16.1.44
t=0 0
m=image 4400 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxTranscodingMMR
a=T38FaxTranscodingJBIG
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPFEC

---

в ответ получает от linksys:

<--- SIP read from UDP:172.16.1.5:5060 --->
SIP/2.0 200 OK
To: 1104 <sip:1104@172.16.1.44>;tag=598ca75348a42654o0
From: <sip:122@172.16.1.44>;tag=as57e290fa
Call-ID: 6a928e7f-dafd8918@172.16.1.5
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.16.1.44:5060;branch=z9hG4bK09d13e17
Contact: 1104 <sip:1104@172.16.1.5:5060>
Server: Linksys/SPA2102-5.2.10
Content-Length: 263
Content-Type: application/sdp

v=0
o=- 7941189 7941189 IN IP4 172.16.1.5
s=-
c=IN IP4 172.16.1.5
t=0 0
m=image 16412 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
--- (10 headers 12 lines) ---

и они они отлично переключаются на t.38.
А при общении 2-х * принимающая сторона говорит о готовности принимать по t.38:

set_destination: Parsing <sip:asterisk2@172.16.1.44> for address/port to send to
set_destination: set destination to 172.16.1.44, port 5060
Reliably Transmitting (no NAT) to 172.16.1.44:5060:
INVITE sip:asterisk2@172.16.1.44 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK635f7e06;rport
Max-Forwards: 70
From: <sip:1104@172.16.1.4>;tag=as30da0557
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as0bca8722
Contact: <sip:1104@172.16.1.4>
Call-ID: 3b09ba4a1fdb616e73d9f5154bfe6a69@172.16.1.44
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 334070128 334070129 IN IP4 172.16.1.4
s=Asterisk PBX 1.6.2.2
c=IN IP4 172.16.1.4
t=0 0
m=image 4204 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxTranscodingMMR
a=T38FaxTranscodingJBIG
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPFEC

---

а передающая сторона не желает общаться по t.38 и передача идет по аудио каналу:

<--- SIP read from UDP:172.16.1.44:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.4:5060;branch=z9hG4bK635f7e06;received=172.16.1.4;rport=5060
From: <sip:1104@172.16.1.4>;tag=as30da0557
To: "asterisk" <sip:asterisk2@172.16.1.44>;tag=as0bca8722
Call-ID: 3b09ba4a1fdb616e73d9f5154bfe6a69@172.16.1.44
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk2@172.16.1.44>
Content-Length: 0


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


Может, кто-нибудь подскажет, где копать?