проблема при приеме факса (free fax for asterisk)

[Vlad1983]

покажите дамп сигналки пира gts-sip

[ded]

Не сдаваться и разбираться - это не значит писать на форум "Помогите! Не работает!"
Чтобы принять в ReceiveFAX по Т38 думаю нужно T38modem, а к нему всё тот же Hylafax. Это не простое решение.

[forik]

покажите дамп сигналки пира gts-sip

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from UDP:89.232.125.48:5060 --->
INVITE sip:s@91.225.76.2 SIP/2.0
From: <sip:8432920231@tattele.com;user=phone>;tag=14107
To: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 980 INVITE
Via: SIP/2.0/UDP 89.232.125.48:5060;branch=z9hG4bK-de758-364fb174-5a8c95d2
content-type: application/sdp
contact: <sip:8432920231@89.232.125.48:5060;user=phone>
user-agent: Nortel SESM 12.0.6.12
max-forwards: 20
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption,replaces,100rel
remote-party-id: <sip:8432920231@tattele.com;user=phone>;screen=yes;party=calling;npi=NPI_E164;ton=TON_NATIONAL
p-asserted-identity: <sip:8432920231@89.232.125.48>
allow: UPDATE
x-nt-corr-id: 5761.004-12-44-56-96@022
x-nt-location: -1
x-nt-service: zoneid=0
Content-Length: 200

v=0
o=PVG 1320745365910 1320745365910 IN IP4 89.232.125.148
s=-
p=+1 6135555555
t=0 0
m=audio 43726 RTP/AVP 8 13 101
c=IN IP4 89.232.125.148
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (18 headers 9 lines) ---
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
Sending to 89.232.125.48 : 5060 (no NAT)
Using INVITE request as basis request - 20159395_1338290f25f@SSLI0
Found peer 'gts-sip' for '8432920231' from 89.232.125.48:5060
Found RTP audio format 8
Found RTP audio format 13
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Peer audio RTP is at port 89.232.125.148:43726
Looking for s in gts-out (domain 91.225.76.2)
list_route: hop: <sip:8432920231@89.232.125.48:5060;user=phone>

<--- Transmitting (no NAT) to 89.232.125.48:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.232.125.48:5060;branch=z9hG4bK-de758-364fb174-5a8c95d2;received=89.232.125.48
From: <sip:8432920231@tattele.com;user=phone>;tag=14107
To: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 980 INVITE
Server: SkynetKazan
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:s@91.225.76.2>
Content-Length: 0


<------------>
-- Executing [s@gts-out:1] Goto("SIP/gts-sip-00000026", "fax-rx,fax,1") in new stack
-- Goto (fax-rx,fax,1)
-- Executing [fax@fax-rx:1] NoOp("SIP/gts-sip-00000026", "***RECEIVING FAX***") in new stack
-- Executing [fax@fax-rx:2] Set("SIP/gts-sip-00000026", "GLOBAL(FAXCOUNT)=4") in new stack
== Setting global variable 'FAXCOUNT' to '4'
-- Executing [fax@fax-rx:3] Set("SIP/gts-sip-00000026", "GLOBAL(FAXCOUNT)=5") in new stack
== Setting global variable 'FAXCOUNT' to '5'
-- Executing [fax@fax-rx:4] Set("SIP/gts-sip-00000026", "FAXCOUNT=5") in new stack
-- Executing [fax@fax-rx:5] Set("SIP/gts-sip-00000026", "FAXFILE=fax-5-rx.tif") in new stack
-- Executing [fax@fax-rx:6] Set("SIP/gts-sip-00000026", "FAXOPT(ecm)=yes") in new stack
-- Executing [fax@fax-rx:7] Set("SIP/gts-sip-00000026", "FAXOPT(headerinfo)=MY FAXBACK RX") in new stack
-- Executing [fax@fax-rx:8] Set("SIP/gts-sip-00000026", "FAXOPT(localstationid)=88432920231") in new stack
-- Executing [fax@fax-rx:9] Set("SIP/gts-sip-00000026", "FAXOPT(maxrate)=14400") in new stack
-- Executing [fax@fax-rx:10] Set("SIP/gts-sip-00000026", "FAXOPT(minrate)=2400") in new stack
-- Executing [fax@fax-rx:11] ReceiveFAX("SIP/gts-sip-00000026", "/var/spool/asterisk/tmp/fax-5-rx.tif") in new stack
-- Channel 'SIP/gts-sip-00000026' receiving FAX '/var/spool/asterisk/tmp/fax-5-rx.tif'
Audio is at 91.225.76.2 port 17654
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 89.232.125.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.232.125.48:5060;branch=z9hG4bK-de758-364fb174-5a8c95d2;received=89.232.125.48
From: <sip:8432920231@tattele.com;user=phone>;tag=14107
To: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 980 INVITE
Server: SkynetKazan
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:s@91.225.76.2>
Content-Type: application/sdp
Content-Length: 245

v=0
o=root 2115957292 2115957292 IN IP4 91.225.76.2
s=Asterisk PBX 1.6.2.9-2ubuntu2.1
c=IN IP4 91.225.76.2
t=0 0
m=audio 17654 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:89.232.125.48:5060 --->
ACK sip:s@91.225.76.2 SIP/2.0
From: <sip:8432920231@tattele.com;user=phone>;tag=14107
To: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 980 ACK
Via: SIP/2.0/UDP 89.232.125.48:5060;branch=z9hG4bK-de758-364fb18a-57b42cf3
contact: <sip:8432920231@89.232.125.48:5060;user=phone>
user-agent: Nortel SESM 12.0.6.12
max-forwards: 20
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:8432920231@89.232.125.48:5060;user=phone> for address/port to send to
set_destination: set destination to 89.232.125.48, port 5060
Reliably Transmitting (no NAT) to 89.232.125.48:5060:
INVITE sip:8432920231@89.232.125.48:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 91.225.76.2:5060;branch=z9hG4bK28b55270;rport
Max-Forwards: 70
From: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
To: <sip:8432920231@tattele.com;user=phone>;tag=14107
Contact: <sip:s@91.225.76.2>
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 102 INVITE
User-Agent: SkynetKazan
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
upported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 2115957292 2115957293 IN IP4 91.225.76.2
s=Asterisk PBX 1.6.2.9-2ubuntu2.1
c=IN IP4 91.225.76.2
t=0 0
m=image 4205 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

---

<--- SIP read from UDP:89.232.125.48:5060 --->
SIP/2.0 100 Trying
From: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
To: <sip:8432920231@tattele.com;user=phone>;tag=14107
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 102 INVITE
Via: SIP/2.0/UDP 91.225.76.2:5060;rport=5060;branch=z9hG4bK28b55270
Content-Length: 0


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

<--- SIP read from UDP:89.232.125.48:5060 --->
SIP/2.0 488 Not Acceptable Here
From: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
To: <sip:8432920231@tattele.com;user=phone>;tag=14107
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 102 INVITE
v: SIP/2.0/UDP 91.225.76.2:5060;rport=5060;branch=z9hG4bK28b55270
user-agent: Nortel SESM 12.0.6.12
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:8432920231@89.232.125.48:5060;user=phone> for address/port to send to
set_destination: set destination to 89.232.125.48, port 5060
Transmitting (no NAT) to 89.232.125.48:5060:
ACK sip:8432920231@89.232.125.48:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 91.225.76.2:5060;branch=z9hG4bK28b55270;rport
Max-Forwards: 70
From: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
To: <sip:8432920231@tattele.com;user=phone>;tag=14107
Contact: <sip:s@91.225.76.2>
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 102 ACK
User-Agent: SkynetKazan
Content-Length: 0


---
set_destination: Parsing <sip:8432920231@89.232.125.48:5060;user=phone> for address/port to send to
set_destination: set destination to 89.232.125.48, port 5060
Audio is at 91.225.76.2 port 17654
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 89.232.125.48:5060:
INVITE sip:8432920231@89.232.125.48:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 91.225.76.2:5060;branch=z9hG4bK279e939a;rport
Max-Forwards: 70
From: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
To: <sip:8432920231@tattele.com;user=phone>;tag=14107
Contact: <sip:s@91.225.76.2>
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 103 INVITE
User-Agent: SkynetKazan
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: 245

v=0
o=root 2115957292 2115957294 IN IP4 91.225.76.2
s=Asterisk PBX 1.6.2.9-2ubuntu2.1
c=IN IP4 91.225.76.2
t=0 0
m=audio 17654 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Nov 8 12:45:00] WARNING[2952]: res_fax.c:1387 receivefax_t38_init: channel 'SIP/gts-sip-00000026' refused to negotiate T.38
[Nov 8 12:45:00] WARNING[2952]: res_fax.c:1408 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/gts-sip-00000026' and T.38 negotiation failed; aborting.
[Nov 8 12:45:00] ERROR[2952]: res_fax.c:1612 receivefax_exec: error initializing channel 'SIP/gts-sip-00000026' in T.38 mode
== Spawn extension (fax-rx, fax, 11) exited non-zero on 'SIP/gts-sip-00000026'
Scheduling destruction of SIP dialog '20159395_1338290f25f@SSLI0' in 32000 ms (Method: ACK)
[Nov 8 12:45:00] ERROR[2952]: cdr_csv.c:306 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied

<--- SIP read from UDP:89.232.125.48:5060 --->
SIP/2.0 100 Trying
From: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
To: <sip:8432920231@tattele.com;user=phone>;tag=14107
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 103 INVITE
Via: SIP/2.0/UDP 91.225.76.2:5060;rport=5060;branch=z9hG4bK279e939a
Content-Length: 0


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

<--- SIP read from UDP:89.232.125.48:5060 --->
SIP/2.0 200 OK
From: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
To: <sip:8432920231@tattele.com;user=phone>;tag=14107
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 103 INVITE
v: SIP/2.0/UDP 91.225.76.2:5060;rport=5060;branch=z9hG4bK279e939a
content-type: application/sdp
contact: <sip:8432920231@89.232.125.48:5060;user=phone>
user-agent: Nortel SESM 12.0.6.12
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption
x-nt-location: -1
Content-Length: 197

v=0
o=PVG 1320745369270 1320745365911 IN IP4 89.232.125.148
s=-
p=+1 6135555555
t=0 0
m=audio 52774 RTP/AVP 8 101
c=IN IP4 89.232.125.148
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 89.232.125.148:52774
set_destination: Parsing <sip:8432920231@89.232.125.48:5060;user=phone> for address/port to send to
set_destination: set destination to 89.232.125.48, port 5060
Transmitting (no NAT) to 89.232.125.48:5060:
ACK sip:8432920231@89.232.125.48:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 91.225.76.2:5060;branch=z9hG4bK24d50a6c;rport
Max-Forwards: 70
From: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
To: <sip:8432920231@tattele.com;user=phone>;tag=14107
Contact: <sip:s@91.225.76.2>
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 103 ACK
User-Agent: SkynetKazan
Content-Length: 0


---
set_destination: Parsing <sip:8432920231@89.232.125.48:5060;user=phone> for address/port to send to
set_destination: set destination to 89.232.125.48, port 5060
Reliably Transmitting (no NAT) to 89.232.125.48:5060:
BYE sip:8432920231@89.232.125.48:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 91.225.76.2:5060;branch=z9hG4bK4da19d87;rport
Max-Forwards: 70
From: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
To: <sip:8432920231@tattele.com;user=phone>;tag=14107
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 104 BYE
User-Agent: SkynetKazan
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0


---
Scheduling destruction of SIP dialog '20159395_1338290f25f@SSLI0' in 32000 ms (Method: ACK)

<--- SIP read from UDP:89.232.125.48:5060 --->
SIP/2.0 200 OK
From: "SIPLineUser SIPLineUser"<sip:5620640@tattele.com>;tag=as7c7b109e
To: <sip:8432920231@tattele.com;user=phone>;tag=14107
Call-ID: 20159395_1338290f25f@SSLI0
CSeq: 104 BYE
Via: SIP/2.0/UDP 91.225.76.2:5060;rport=5060;branch=z9hG4bK4da19d87
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '20159395_1338290f25f@SSLI0' Method: ACK

[forik]

Это не простое решение.

Мне всего-то надо входящие факсы на мыло отправить. Подскажите пожалйста как это тогда можно сделать по-простому?

[Vlad1983]

что в фаерволе понакручено касательно портов в районе 4205 (диапазона из udptl.conf)?

и что-то мне не нравится после SDP c t38 ответ
SIP/2.0 488 Not Acceptable Here
оператор точно поддерживает t38?

[ded]

Мне всего-то надо входящие факсы на мыло отправить. Подскажите пожалйста как это тогда можно сделать по-простому?

Всего-то?
Это никогда ни у кого не было по-простому.

Влад, <--- SIP read from UDP:89.232.125.48:5060 --->
SIP/2.0 488 Not Acceptable Here
от Нортеля приходит именно на предложение Астериска о Т38.

[forik]

В фаерволе все открыто в общем-то (его вообще нет), сам asterisk использует:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

udp 0 0 *:5000 *:* 857/asterisk
udp 0 0 *:4520 *:* 857/asterisk
udp 0 0 *:sip *:* 857/asterisk

[Vlad1983]

стоит попробовать отключить t38 и покидать голосом.
убрать faxdetect=yes

[forik]

убрать faxdetect=yes

У провайдера уточнил, т38 в теории поддерживается.
Попробовал по g711 и заработало! Но как-то изображение сжимается, читабельно, но не айс.
В любом случае это сдвиг с мертвой точки ) Спасибо огромное, Vlad1983, ded!

Изображение квадратным становится, странность какая-то..

[zzuz]

Но как-то изображение сжимается, читабельно, но не айс.

А если в pdf сконвертировать?