Прошу помощи у гуру.

Я пытаюсь отправить факс с сервера при помощи SendFax . При копировании колл файла в /home/asterisk/var/asterisk/spool все происходит как задуманно, астериск звонит на указанный номер (там сидит факс с автоприемом) который отвечает и тут все :(

Астериск находится за фаерволом на котором промапленны все нужные порты (RTP, UDPTL, 5060).

Логи и конфиги ниже:


****
/home/asterisk/sbin/asterisk -V
Asterisk SVN-trunk-r258685


*** my.call

Channel: SIP/sipprov/xxxxxxxxxx
Callerid: yyyyyyyyyyy
MaxRetries: 3
RetryTime: 10
WaitTime: 25
Context: fax_out
Extension: faxout1
Set: FAXFILE=/home/sip/fax.tif

*** sip.conf

[sipprov]
type=peer
host=sip.provider.net
canreinvite=no
dtmfmode=rfc2833
t38pt_udptl=yes
qualify=yes
nat=yes


*** Extensions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo

[default]

[fax_out]
exten => faxout1,1,Answer()
exten => faxout1,n,Wait(2)
exten => faxout1,n,Playback(vm-goodbye)
exten => faxout1,n,SendFAX(${FAXFILE})
exten => faxout1,n,Hangup


**** Asterisk Log:

*CLI> -- Attempting call on SIP/sipprov/XXXXXXXX for faxout1@fax_out:1 (Retry 1)
> Channel SIP/sipprov-00000000 was answered.
-- Executing [faxout1@fax_out:1] Answer("SIP/sipprov-00000000", "") in new stack
-- Executing [faxout1@fax_out:2] Wait("SIP/sipprov-00000000", "2") in new stack
-- Executing [faxout1@fax_out:3] Playback("SIP/sipprov-00000000", "vm-goodbye") in new stack
-- <SIP/sipprov-00000000> Playing 'vm-goodbye.gsm' (language 'en')
-- Executing [faxout1@fax_out:4] SendFAX("SIP/sipprov-00000000", "/home/sip/fax.tif") in new stack
[Apr 29 13:49:11] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:11] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:16] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:16] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:16] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:16] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:24] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:24] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:24] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:24] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:32] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:32] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:32] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:32] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:39] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:39] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:40] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:49:40] WARNING[30896]: res_rtp_asterisk.c:1912 ast_rtp_read: RTP Read too short
[Apr 29 13:50:11] WARNING[30896]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=2: Timed out waiting for initial communication.
[Apr 29 13:50:11] WARNING[30896]: app_fax.c:823 transmit: Transmission failed
-- Executing [faxout1@fax_out:5] Hangup("SIP/sipprov-00000000", "") in new stack
== Spawn extension (fax_out, faxout1, 5) exited non-zero on 'SIP/sipprov-00000000'
[Apr 29 13:50:11] NOTICE[30896]: pbx_spool.c:352 attempt_thread: Call completed to SIP/sipprov/XXXXXXXX
[Apr 29 13:50:18] WARNING[30892]: pbx_spool.c:394 scan_service: Unable to open /home/asterisk/var/spool/asterisk/outgoing/my.call: No such file or directory, deleting
[Apr 29 13:50:18] WARNING[30892]: pbx_spool.c:394 scan_service: Unable to open /home/asterisk/var/spool/asterisk/outgoing/my.call: No such file or directory, deleting
[Apr 29 13:50:18] WARNING[30892]: pbx_spool.c:394 scan_service: Unable to open /home/asterisk/var/spool/asterisk/outgoing/my.call: No such file or directory, deleting
[Apr 29 13:50:18] WARNING[30892]: pbx_spool.c:394 scan_service: Unable to open /home/asterisk/var/spool/asterisk/outgoing/my.call: No such file or directory, deleting

Прям таки раз и T38 через SendFax ?

zzuz: нипанимаю сути уточняющего вопроса по теме :( может есть более дельный совет?

zzuz имеет ввиду, что астерисковый senfax не умеет t38

Нет . Вопрос в том поддерживает ли удаленная сторона T38. Я лично не пробывал t38 с халявным модулем( если конечно тут халявный FFA).
1.fax set debug on
2.core set verbose 6

в логе отчетливо видно , что кто-ждет t38udptl, а кто-то 711. И раз уж так , то для пира укажите allow=alaw. без поллитра не разберешься... то есть без дебага)

Подскажите в чем может быть проблема, не принимает/отправляет факсы, после того как перешли на 1.6 трейс не успешного вызова. АТС-mediant2000-asterisk-dlink DVG 5402ps
Если напрямую без asterisk факсы ходят.
<--- SIP read from UDP://10.8.108.234:5060 --->
INVITE sip:670007@192.168.58.246:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bKef2f9970903b6c36
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>
Call-ID:1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq:41 INVITE
Contact:<sip:675401@10.8.108.234:5060>
Expires:90
Max-Forwards:70
Supported:replaces
User-Agent:dlink 12-37-55916142-0.8.22.1
Content-Type:application/sdp
Content-Length:259

v=0
o=675401 1802247060 1802247060 IN IP4 10.8.108.234
s=Session SDP
c=IN IP4 10.8.108.234
t=0 0
m=audio 9000 RTP/AVP 18 4 2 0 8
a=rtpmap:18 G729/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:2 G726-32/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1

<------------->
--- (14 headers 11 lines) ---
Sending to 10.8.108.234 : 5060 (no NAT)
Using INVITE request as basis request - 1B19-7DFB-46695052B48B4827186E-022@SipHost
Found user '675401' for '675401'
aster*CLI>
<--- Reliably Transmitting (no NAT) to 10.8.108.234:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bKef2f9970903b6c36;received=10.8.108.234
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>;tag=as33b1e8cf
Call-ID: 1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq: 41 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6bfc4c97"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1B19-7DFB-46695052B48B4827186E-022@SipHost' in 32000 ms (Method: INVITE)
aster*CLI>
<--- SIP read from UDP://10.8.108.234:5060 --->
ACK sip:670007@192.168.58.246:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bKef2f9970903b6c36
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>;tag=as33b1e8cf
Call-ID:1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq:41 ACK
Max-Forwards:70
Content-Length:0


<------------->
--- (8 headers 0 lines) ---
aster*CLI>
<--- SIP read from UDP://10.8.108.234:5060 --->
INVITE sip:670007@192.168.58.246:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bKa7af5ddefd3decd1
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>
Call-ID:1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq:42 INVITE
Contact:<sip:675401@10.8.108.234:5060>
Expires:90
Max-Forwards:70
Authorization:Digest username="675401",realm="asterisk",nonce="6bfc4c97",uri="sip:670007@192.168.58.246:5060;user=phone",response="710a8c4d07bba8e64dabd511fde4386e",algorithm=MD5
Supported:replaces
User-Agent:dlink 12-37-55916142-0.8.22.1
Content-Type:application/sdp
Content-Length:259

v=0
o=675401 1802247060 1802247060 IN IP4 10.8.108.234
s=Session SDP
c=IN IP4 10.8.108.234
t=0 0
m=audio 9000 RTP/AVP 18 4 2 0 8
a=rtpmap:18 G729/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:2 G726-32/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1

<------------->
--- (15 headers 11 lines) ---
Sending to 10.8.108.234 : 5060 (no NAT)
Using INVITE request as basis request - 1B19-7DFB-46695052B48B4827186E-022@SipHost
Found user '675401' for '675401'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 0
Found RTP audio format 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format G726-32 for ID 2
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.8.108.234:9000
Looking for 670007 in mezhblock (domain 192.168.58.246)
list_route: hop: <sip:675401@10.8.108.234:5060>
aster*CLI>
<--- Transmitting (no NAT) to 10.8.108.234:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bKa7af5ddefd3decd1;received=10.8.108.234
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>
Call-ID: 1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq: 42 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:670007@192.168.58.246>
Content-Length: 0


<------------>
-- Executing [670007@mezhblock:1] Dial("SIP/675401-0000031f", "SIP/670007@mediant") in new stack
-- Called 670007@mediant
-- SIP/mediant-00000320 is making progress passing it to SIP/675401-0000031f
Audio is at 192.168.58.246 port 19762
Adding codec 0x8 (alaw) to SDP

<--- Transmitting (no NAT) to 10.8.108.234:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bKa7af5ddefd3decd1;received=10.8.108.234
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>;tag=as73a26123
Call-ID: 1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq: 42 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:670007@192.168.58.246>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 1605670641 1605670641 IN IP4 192.168.58.246
s=Asterisk PBX 1.6.0.21
c=IN IP4 192.168.58.246
t=0 0
m=audio 19762 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
-- SIP/mediant-00000320 is ringing
aster*CLI>
<--- Transmitting (no NAT) to 10.8.108.234:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bKa7af5ddefd3decd1;received=10.8.108.234
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>;tag=as73a26123
Call-ID: 1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq: 42 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:670007@192.168.58.246>
Content-Length: 0


<------------>
-- SIP/mediant-00000320 is making progress passing it to SIP/675401-0000031f
== Spawn extension (tosub, 675313, 11) exited non-zero on 'SIP/mediant-0000031d'
> Saved useragent "dlink 12-38-37912445-0.10.33.1-DSLX" for peer 675590
> Saved useragent "dlink 12-38-37912445-0.10.33.1-DSLX" for peer 675508
-- SIP/mediant-00000320 answered SIP/675401-0000031f
Audio is at 192.168.58.246 port 19762
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 10.8.108.234:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bKa7af5ddefd3decd1;received=10.8.108.234
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>;tag=as73a26123
Call-ID: 1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq: 42 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:670007@192.168.58.246>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 1605670641 1605670642 IN IP4 192.168.58.246
s=Asterisk PBX 1.6.0.21
c=IN IP4 192.168.58.246
t=0 0
m=audio 19762 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
-- Packet2Packet bridging SIP/675401-0000031f and SIP/mediant-00000320
aster*CLI>
<--- SIP read from UDP://10.8.108.234:5060 --->
ACK sip:670007@192.168.58.246 SIP/2.0
Via:SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bK5b6e6f64367cd185
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>;tag=as73a26123
Call-ID:1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq:42 ACK
Max-Forwards:70
Authorization:Digest username="675401",realm="asterisk",nonce="6bfc4c97",uri="sip:670007@192.168.58.246:5060;user=phone",response="710a8c4d07bba8e64dabd511fde4386e",algorithm=MD5
User-Agent:dlink 12-37-55916142-0.8.22.1
Content-Length:0


<------------->
--- (10 headers 0 lines) ---
-- Executing [9531905@mezhgorod:1] Dial("SIP/675603-00000321", "SIP/9531905@mediant") in new stack
-- Called 9531905@mediant
-- SIP/mediant-00000322 is making progress passing it to SIP/675603-00000321
> Saved useragent "dlink 12-38-39912586-0.10.37.1-DSLX" for peer 675616
> Saved useragent "dlink 12-38-37912445-0.10.33.1-DSLX" for peer 675342
> Saved useragent "dlink 12-38-37912445-0.10.33.1-DSLX" for peer 675574
set_destination: Parsing <sip:675401@10.8.108.234:5060> for address/port to send to
set_destination: set destination to 10.8.108.234, port 5060
Reliably Transmitting (no NAT) to 10.8.108.234:5060:
INVITE sip:675401@10.8.108.234:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.246:5060;branch=z9hG4bK36e7cbb4;rport
Max-Forwards: 70
From: <sip:670007@192.168.58.246:5060;user=phone>;tag=as73a26123
To: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
Contact: <sip:670007@192.168.58.246>
Call-ID: 1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.21
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: 268

v=0
o=root 1605670641 1605670643 IN IP4 192.168.58.246
s=Asterisk PBX 1.6.0.21
c=IN IP4 192.168.58.246
t=0 0
m=image 4421 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:85
a=T38FaxUdpEC:t38UDPFEC

---
aster*CLI>
<--- SIP read from UDP://10.8.108.234:5060 --->
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.58.246:5060;rport;branch=z9hG4bK36e7cbb4
From: <sip:670007@192.168.58.246:5060;user=phone>;tag=as73a26123
To: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
Call-ID:1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq:102 INVITE
Content-Type:application/sdp
Content-Length:0


<------------->
--- (8 headers 0 lines) ---
aster*CLI>
<--- SIP read from UDP://10.8.108.234:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 192.168.58.246:5060;rport;branch=z9hG4bK36e7cbb4
From: <sip:670007@192.168.58.246:5060;user=phone>;tag=as73a26123
To: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
Call-ID:1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq:102 INVITE
Contact:<sip:675401@10.8.108.234:5060>
User-Agent:dlink 12-37-55916142-0.8.22.1
Content-Type:application/sdp
Content-Length:256

v=0
o=675401 1802261790 1802261790 IN IP4 10.8.108.234
s=Session SDP
c=IN IP4 10.8.108.234
t=0 0
m=image 9000 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:85
a=T38FaxUdpEC:t38UDPFEC

<------------->
--- (11 headers 11 lines) ---
Got T.38 offer in SDP in dialog 1B19-7DFB-46695052B48B4827186E-022@SipHost
Capabilities: us - 0x8 (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.
set_destination: Parsing <sip:675401@10.8.108.234:5060> for address/port to send to
set_destination: set destination to 10.8.108.234, port 5060
Transmitting (no NAT) to 10.8.108.234:5060:
ACK sip:675401@10.8.108.234:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.246:5060;branch=z9hG4bK609deb61;rport
Max-Forwards: 70
From: <sip:670007@192.168.58.246:5060;user=phone>;tag=as73a26123
To: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
Contact: <sip:670007@192.168.58.246>
Call-ID: 1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.21
Content-Length: 0


---
> Saved useragent "dlink 12-38-37912445-0.10.33.1-DSLX" for peer 675540
> Saved useragent "dlink 12-38-37912445-0.10.33.1-DSLX" for peer 675501
-- Executing [9540699@gorod:1] Dial("SIP/675501-00000323", "SIP/9540699@mediant") in new stack
-- Called 9540699@mediant
-- SIP/mediant-00000324 is making progress passing it to SIP/675501-00000323
aster*CLI>
<--- SIP read from UDP://10.8.108.234:5060 --->
REGISTER sip:192.168.58.246:5060 SIP/2.0
Via:SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bK2202f1f92ec95fa6
From: "675401" <sip:675401@192.168.58.246>;tag=bdf42318-695074
To: "675401" <sip:675401@192.168.58.246>
Call-ID:1B19-7DFB-46695074B322B27DD049-023@SipHost
CSeq:43 REGISTER
Contact:<sip:675401@10.8.108.234:5060>
Expires:600
Max-Forwards:70
User-Agent:dlink 12-37-55916142-0.8.22.1
Content-Length:0


<------------->
--- (11 headers 0 lines) ---
Sending to 10.8.108.234 : 5060 (no NAT)
aster*CLI>
<--- Transmitting (no NAT) to 10.8.108.234:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bK2202f1f92ec95fa6;received=10.8.108.234
From: "675401" <sip:675401@192.168.58.246>;tag=bdf42318-695074
To: "675401" <sip:675401@192.168.58.246>;tag=as02f311a8
Call-ID: 1B19-7DFB-46695074B322B27DD049-023@SipHost
CSeq: 43 REGISTER
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54ac08a2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1B19-7DFB-46695074B322B27DD049-023@SipHost' in 32000 ms (Method: REGISTER)
aster*CLI>
<--- SIP read from UDP://10.8.108.234:5060 --->
REGISTER sip:192.168.58.246:5060 SIP/2.0
Via:SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bKcaf68b277485fe58
From: "675401" <sip:675401@192.168.58.246>;tag=bdf42318-695074
To: "675401" <sip:675401@192.168.58.246>
Call-ID:1B19-7DFB-46695074B322B27DD049-023@SipHost
CSeq:44 REGISTER
Contact:<sip:675401@10.8.108.234:5060>
Expires:600
Max-Forwards:70
Authorization:Digest username="675401",realm="asterisk",nonce="54ac08a2",uri="sip:192.168.58.246:5060",response="e4132be7488fd00e65666eee6395b077",algorithm=MD5
User-Agent:dlink 12-37-55916142-0.8.22.1
Content-Length:0


<------------->
--- (12 headers 0 lines) ---
Sending to 10.8.108.234 : 5060 (no NAT)
> Saved useragent "dlink 12-37-55916142-0.8.22.1" for peer 675401
aster*CLI>
<--- Transmitting (no NAT) to 10.8.108.234:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bKcaf68b277485fe58;received=10.8.108.234
From: "675401" <sip:675401@192.168.58.246>;tag=bdf42318-695074
To: "675401" <sip:675401@192.168.58.246>;tag=as02f311a8
Call-ID: 1B19-7DFB-46695074B322B27DD049-023@SipHost
CSeq: 44 REGISTER
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 600
Contact: <sip:675401@10.8.108.234:5060>;expires=600
Date: Tue, 01 Jun 2010 09:00:14 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1B19-7DFB-46695074B322B27DD049-023@SipHost' in 32000 ms (Method: REGISTER)
> Saved useragent "dlink 12-38-32912305-0.10.26.1-DSLX" for peer 675659
-- SIP/mediant-00000324 is ringing
-- SIP/mediant-00000324 is making progress passing it to SIP/675501-00000323
> Saved useragent "dlink 12-38-37912445-0.10.33.1-DSLX" for peer 675378
== Spawn extension (mezhgorod, 445517, 1) exited non-zero on 'SIP/675623-00000313'
== Spawn extension (mezhgorod, 9531905, 1) exited non-zero on 'SIP/675603-00000321'
== Spawn extension (tosub, 675646, 11) exited non-zero on 'SIP/mediant-00000311'
> Saved useragent "dlink 12-38-39912586-0.10.37.1-DSLX" for peer 675604
-- Executing [89222501072@mezhgorod:1] Dial("SIP/675627-00000325", "SIP/89222501072@mediant") in new stack
-- Called 89222501072@mediant
-- SIP/mediant-00000326 is making progress passing it to SIP/675627-00000325
aster*CLI>
<--- SIP read from UDP://10.8.108.234:5060 --->
BYE sip:670007@192.168.58.246 SIP/2.0
Via:SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bK14316e98a6fd629a
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>;tag=as73a26123
Call-ID:1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq:45 BYE
Max-Forwards:70
User-Agent:dlink 12-37-55916142-0.8.22.1
Content-Length:0


<------------->
--- (9 headers 0 lines) ---
Sending to 10.8.108.234 : 5060 (no NAT)

<--- Transmitting (no NAT) to 10.8.108.234:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.108.234:5060;branch=z9hG4bK14316e98a6fd629a;received=10.8.108.234
From: "675401" <sip:675401@192.168.58.246>;tag=3eb3b07c-695052
To: <sip:670007@192.168.58.246:5060;user=phone>;tag=as73a26123
Call-ID: 1B19-7DFB-46695052B48B4827186E-022@SipHost
CSeq: 45 BYE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
== Spawn extension (mezhblock, 670007, 1) exited non-zero on 'SIP/675401-0000031f'
Really destroying SIP dialog '1B19-7DFB-46695052B48B4827186E-022@SipHost' Method: BYE

Надо смотреть медиа-сессию. С точки зрения SIP, на первый взгляд, сессия устанавливается и разрывается корректно.

http://file.qip.ru/file/130798606/4f580389/trace.html
Судя по трейсу астериск шлет t38:v21:HDLC:Digital Identification Signal, но длинк не отвечает на него. Когда на прямую без астериска включаем, на DIS от астериска длинк отвечает DCS.

Посмотрели трейс между астериском и медиантом, астериск зачем то шлет еще один лишний инвайт.
|Time | 192.168.58.246 | 192.168.58.15 |
|49,407 | INVITE SDP ( g711A g729 telephone-event) |SIP From: sip:675404@192.168.58.246 To:sip:670007@192.168.58.15:5060
| |(5060) ------------------> (5060) |
|49,430 | 100 Trying| |SIP Status
| |(5060) <------------------ (5060) |
|49,480 | 183 Session Progress SDP ( g711A telephone-eve... |SIP Status
| |(5060) <------------------ (5060) |
|49,502 | RTP (g711A) |RTP Num packets:513 Duration:10.236s SSRC:0x328F12D4
| |(17822) <------------------ (6570) |
|49,525 | RTP (g711A) |RTP Num packets:1 Duration:0.000s SSRC:0x43F2659F
| |(17822) ------------------> (6570) |
|49,525 | 180 Ringing SDP ( g711A telephone-event) |SIP Status
| |(5060) <------------------ (5060) |
|49,502 | RTP (g711A) |RTP Num packets:135 Duration:2.680s SSRC:0x328F12D4
| |(17822) <------------------ (6570) |
|49,546 | RTP (g711A) |RTP Num packets:377 Duration:7.519s SSRC:0x43F2659F
| |(17822) ------------------> (6570) |
|57,075 | 200 OK SDP ( g711A telephone-event) |SIP Status
| |(5060) <------------------ (5060) |
|57,075 | ACK | |SIP Request
| |(5060) ------------------> (5060) |
|57,084 | RTP (g711A) |RTP Num packets:337 Duration:6.721s SSRC:0x4121C28B
| |(17822) ------------------> (6570) |
|63,776 | INVITE SDP ( t38) |SIP Request
| |(5060) <------------------ (5060) |
|63,781 | 100 Trying| |SIP Status
| |(5060) ------------------> (5060) |
|63,821 | 200 OK SDP ( t38) |SIP Status
| |(5060) ------------------> (5060) |
|63,848 | ACK | |SIP Request
| |(5060) <------------------ (5060) |
|65,200 | cng | |t38:t30 Ind:cng
| |(4010) ------------------> (6572) |
|66,345 | no-signal | |t38:t30 Ind:no-signal
| |(4010) <------------------ (6572) |
|68,730 | cng | |t38:t30 Ind:cng
| |(4010) ------------------> (6572) |
|69,645 | v21-preamble |t38:t30 Ind:v21-preamble
| |(4010) <------------------ (6572) |
|70,782 | INVITE SDP ( g711A telephone-event) |SIP From: sip:675404@192.168.58.246 To:sip:670007@192.168.58.15:5060
| |(5060) ------------------> (5060) |
|70,806 | 200 OK SDP ( g711A telephone-event) |SIP Status
| |(5060) <------------------ (5060) |
|70,806 | ACK | |SIP Request
| |(5060) ------------------> (5060) |
|70,870 | RTP (g711A) |RTP Num packets:720 Duration:14.379s SSRC:0x3A15378D
| |(17822) ------------------> (6570) |
|70,896 | RTP (g711A) |RTP Num packets:350 Duration:6.980s SSRC:0x1C4203A9
| |(17822) <------------------ (6570) |
|85,250 | BYE | |SIP Request
| |(5060) <------------------ (5060) |
|85,251 | 200 OK | |SIP Status
| |(5060) ------------------> (5060) |

Вот и славно, вот и определили проблему. Теперь вам осталось выяснить разницу в двух DIS'ах и все, вопрос закрыт.