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Free Fax for Asterisk неработает ReciveFax

Проблемы Asterisk без вэб-оболочек и их решения

Модераторы: april22, Zavr2008

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Morya4ok-cr
Сообщения: 6
Зарегистрирован: 10 окт 2012, 19:04

Free Fax for Asterisk неработает ReciveFax

Сообщение Morya4ok-cr »

Всем хорошего настроения!
У меня проблема с Free Fax for Asterisk не получается получить факс.
Asterisk 10 FreePbx 2.10
Скачал модуль, поставил, зарегистрировал в консоли появились модули

Код: Выделить всё

Module                         Description                              Use Count
res_fax.so                     Generic FAX Applications                 1
res_fax_digium.so              Digium G.711 and T.38 FAX Technologies ( 0
вот в диалплане пишу следующее:

Код: Выделить всё

[infax]
exten => receive,1,NoOp(**** FAX RECEIVE ****)
exten => receive,n,Set(GLOBAL(FAXCOUNT)=$[${SHELL(ls -1 /var/www/html/in_fax/*.tif | wc -l)} +1])
exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten => receive,n,Set(FAXFILE=fax-${CALLERID(num)}-${FAXCOUNT}.tif)
exten => receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)})
exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
exten => receive,n,NoOp(**** SETTING FAXOPT ****)
exten => receive,n,Set(FAXOPT(ecm)=yes)
exten => receive,n,Set(FAXOPT(headerinfo)=MY FAXBACK RX)
exten => receive,n,Set(FAXOPT(localstationid)=875)
exten => receive,n,Set(FAXOPT(maxrate)=14400)
exten => receive,n,Set(FAXOPT(minrate)=2400)
exten => receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten => receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten => receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten => receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten => receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten => receive,n,NoOp(**** RECEIVING FAX : ${FAXFILE} ****)
exten => receive,n,ReceiveFAX(/var/www/html/in_fax/${FAXFILE},f)
Если набираю номер виртуального факса то гудки факса я слышу дальше
при попытке отправить с моего локального факса, на этот виртуальный, на факсе выдает ошибку "не удалось установить соединение"
после того как включил т38
повесился *

Код: Выделить всё

-- Executing [875@from-internal:1] Goto("SIP/855-0000130f", "infax,receive,1") in new stack
    -- Goto (infax,receive,1)
    -- Executing [receive@infax:1] NoOp("SIP/855-0000130f", "**** FAX RECEIVE ****") in new stack
    -- Executing [receive@infax:2] Set("SIP/855-0000130f", "GLOBAL(FAXCOUNT)=1") in new stack
  == Setting global variable 'FAXCOUNT' to '1'
    -- Executing [receive@infax:3] Set("SIP/855-0000130f", "FAXCOUNT=1") in new stack
    -- Executing [receive@infax:4] Set("SIP/855-0000130f", "FAXFILE=fax-855-1.tif") in new stack
    -- Executing [receive@infax:5] Set("SIP/855-0000130f", "GLOBAL(LASTFAXCALLERNUM)=855") in new stack
  == Setting global variable 'LASTFAXCALLERNUM' to '855'
    -- Executing [receive@infax:6] Set("SIP/855-0000130f", "GLOBAL(LASTFAXCALLERNAME)=FAX_test") in new stack
  == Setting global variable 'LASTFAXCALLERNAME' to 'FAX_test'
    -- Executing [receive@infax:7] NoOp("SIP/855-0000130f", "**** SETTING FAXOPT ****") in new stack
    -- Executing [receive@infax:8] Set("SIP/855-0000130f", "FAXOPT(ecm)=yes") in new stack
    -- Executing [receive@infax:9] Set("SIP/855-0000130f", "FAXOPT(headerinfo)=MY FAXBACK RX") in new stack
    -- Executing [receive@infax:10] Set("SIP/855-0000130f", "FAXOPT(localstationid)=875") in new stack
    -- Executing [receive@infax:11] Set("SIP/855-0000130f", "FAXOPT(maxrate)=14400") in new stack
    -- Executing [receive@infax:12] Set("SIP/855-0000130f", "FAXOPT(minrate)=2400") in new stack
    -- Executing [receive@infax:13] NoOp("SIP/855-0000130f", "FAXOPT(ecm) : yes") in new stack
    -- Executing [receive@infax:14] NoOp("SIP/855-0000130f", "FAXOPT(headerinfo) : MY FAXBACK RX") in new stack
    -- Executing [receive@infax:15] NoOp("SIP/855-0000130f", "FAXOPT(localstationid) : 875") in new stack
    -- Executing [receive@infax:16] NoOp("SIP/855-0000130f", "FAXOPT(maxrate) : 14400") in new stack
    -- Executing [receive@infax:17] NoOp("SIP/855-0000130f", "FAXOPT(minrate) : 2400") in new stack
    -- Executing [receive@infax:18] NoOp("SIP/855-0000130f", "**** RECEIVING FAX : fax-855-1.tif ****") in new stack
    -- Executing [receive@infax:19] ReceiveFAX("SIP/855-0000130f", "/var/www/html/in_fax/fax-855-1.tif,f") in new stack
    -- Channel 'SIP/855-0000130f' receiving FAX '/var/www/html/in_fax/fax-855-1.tif'
    -- Channel 'SIP/855-0000130f' FAX session '0' started
freepbx*CLI>
Disconnected from Asterisk server
дебаг peer аналогового факса с выключенным t38
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
Reliably Transmitting (NAT) to 194.0.88.214:1250:
OPTIONS sip:855@192.168.0.35:6892 SIP/2.0
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK22919db1;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@195.191.226.132>;tag=as68390d38
To: <sip:855@192.168.0.35:6892>

From: "Unknown" <sip:Unknown@195.191.226.132>;tag=as68390d38
To: <sip:855@192.168.0.35:6892>
Contact: <sip:Unknown@195.191.226.132:5060>
Call-ID: 1e5c21c149695419742214921d3e9a31@195.191.226.132:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(10.0.0)
Date: Wed, 10 Oct 2012 15:20:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK22919db1;rport=5060
From: "Unknown" <sip:Unknown@195.191.226.132>;tag=as68390d38
To: <sip:855@192.168.0.35:6892>;tag=1736735314
Call-ID: 1e5c21c149695419742214921d3e9a31@195.191.226.132:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1e5c21c149695419742214921d3e9a31@195.191.226.132:5060' Method: OPTIONS
freepbx*CLI> fax set t38cap off


T.38 Session Packet Capture now Disabled

> Saved useragent "X-Lite release 1103k stamp 53621" for peer 463
-- Got SIP response 603 "Decline" back from 194.0.88.214:18504

<--- SIP read from UDP:194.0.88.214:1250 --->
INVITE sip:875@freepbx.edi.su SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK1322411270;rport
Route: <sip:freepbx.edi.su:5060;lr>
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
To: <sip:875@freepbx.edi.su>
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 20 INVITE
Contact: "Anonymous" <sip:855@192.168.0.35:6892>
Max-Forwards: 70
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Privacy: id
P-Asserted-Identity: "FAX_test" <sip:855@freepbx.edi.su>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 455

v=0
o=855 8000 8000 IN IP4 192.168.0.35
s=SIP Call
c=IN IP4 192.168.0.35
t=0 0
m=audio 8577 RTP/AVP 8 0 4 18 112 97 102 100 101
a=sendrecv
a=rtpmap:8 PCMA/8000

a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
<------------->
--- (17 headers 20 lines) ---
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 194.0.88.214:1250 (NAT)
Using INVITE request as basis request - 622459119-6892-3@BJC.BGI.A.DF
Found peer '855' for 'anonymous' from 194.0.88.214:1250

<--- Reliably Transmitting (NAT) to 194.0.88.214:1250 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK1322411270;received=194.0.88.214;rport=1250
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
To: <sip:875@freepbx.edi.su>;tag=as2acf10bf
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 20 INVITE
Server: FPBX-2.10.1(10.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7616ece2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '622459119-6892-3@BJC.BGI.A.DF' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:194.0.88.214:1250 --->
ACK sip:875@freepbx.edi.su SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK1322411270;rport
Route: <sip:freepbx.edi.su:5060;lr>
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
To: <sip:875@freepbx.edi.su>;tag=as2acf10bf
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 20 ACK
Content-Length: 0

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

<--- SIP read from UDP:194.0.88.214:1250 --->
INVITE sip:875@freepbx.edi.su SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK522054693;rport
Route: <sip:freepbx.edi.su:5060;lr>
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
To: <sip:875@freepbx.edi.su>
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 21 INVITE
Contact: "Anonymous" <sip:855@192.168.0.35:6892>
Authorization: Digest username="855", realm="asterisk", nonce="7616ece2", uri="sip:875@freepbx.edi.su", response="ee26c0e2d6c8e6a165bb38f22f585286", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Privacy: id

P-Asserted-Identity: "FAX_test" <sip:855@freepbx.edi.su>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 455

v=0
o=855 8000 8000 IN IP4 192.168.0.35
s=SIP Call
c=IN IP4 192.168.0.35
t=0 0
m=audio 8577 RTP/AVP 8 0 4 18 112 97 102 100 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
<------------->
--- (18 headers 20 lines) ---
Sending to 194.0.88.214:1250 (NAT)
Using INVITE request as basis request - 622459119-6892-3@BJC.BGI.A.DF
Found peer '855' for 'anonymous' from 194.0.88.214:1250
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 112
Found RTP audio format 97
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 112
Found audio description format iLBC for ID 97
Found unknown media description format G729E for ID 102
Found unknown media description format AAL2-G726-16 for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.35:8577
Looking for 875 in from-internal (domain freepbx.edi.su)
list_route: hop: <sip:855@192.168.0.35:6892>

<--- Transmitting (NAT) to 194.0.88.214:1250 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK522054693;received=194.0.88.214;rport=1250

rom: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
To: <sip:875@freepbx.edi.su>
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 21 INVITE
Server: FPBX-2.10.1(10.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:875@195.191.226.132:5060>
Content-Length: 0


<------------>
-- Executing [875@from-internal:1] Goto("SIP/855-00000000", "infax,receive,1") in new stack
-- Goto (infax,receive,1)
-- Executing [receive@infax:1] NoOp("SIP/855-00000000", "**** FAX RECEIVE ****") in new stack
-- Executing [receive@infax:2] Set("SIP/855-00000000", "GLOBAL(FAXCOUNT)=1") in new stack
== Setting global variable 'FAXCOUNT' to '1'
-- Executing [receive@infax:3] Set("SIP/855-00000000", "FAXCOUNT=1") in new stack
-- Executing [receive@infax:4] Set("SIP/855-00000000", "FAXFILE=fax-855-1.tif") in new stack
-- Executing [receive@infax:5] Set("SIP/855-00000000", "GLOBAL(LASTFAXCALLERNUM)=855") in new stack
== Setting global variable 'LASTFAXCALLERNUM' to '855'
-- Executing [receive@infax:6] Set("SIP/855-00000000", "GLOBAL(LASTFAXCALLERNAME)=FAX_test") in new stack
== Setting global variable 'LASTFAXCALLERNAME' to 'FAX_test'
-- Executing [receive@infax:7] NoOp("SIP/855-00000000", "**** SETTING FAXOPT ****") in new stack
-- Executing [receive@infax:8] Set("SIP/855-00000000", "FAXOPT(ecm)=yes") in new stack
-- Executing [receive@infax:9] Set("SIP/855-00000000", "FAXOPT(headerinfo)=MY FAXBACK RX") in new stack
-- Executing [receive@infax:10] Set("SIP/855-00000000", "FAXOPT(localstationid)=875") in new stack
-- Executing [receive@infax:11] Set("SIP/855-00000000", "FAXOPT(maxrate)=14400") in new stack
-- Executing [receive@infax:12] Set("SIP/855-00000000", "FAXOPT(minrate)=2400") in new stack
-- Executing [receive@infax:13] NoOp("SIP/855-00000000", "FAXOPT(ecm) : yes") in new stack
-- Executing [receive@infax:14] NoOp("SIP/855-00000000", "FAXOPT(headerinfo) : MY FAXBACK RX") in new stack
-- Executing [receive@infax:15] NoOp("SIP/855-00000000", "FAXOPT(localstationid) : 875") in new stack
-- Executing [receive@infax:16] NoOp("SIP/855-00000000", "FAXOPT(maxrate) : 14400") in new stack
-- Executing [receive@infax:17] NoOp("SIP/855-00000000", "FAXOPT(minrate) : 2400") in new stack
-- Executing [receive@infax:18] NoOp("SIP/855-00000000", "**** RECEIVING FAX : fax-855-1.tif ****") in new stack
-- Executing [receive@infax:19] ReceiveFAX("SIP/855-00000000", "/var/www/html/in_fax/fax-855-1.tif,f") in new stack
-- Channel 'SIP/855-00000000' receiving FAX '/var/www/html/in_fax/fax-855-1.tif'
Audio is at 11932
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK522054693;received=194.0.88.214;rport=1250
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
To: <sip:875@freepbx.edi.su>;tag=as779d8908
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 21 INVITE
Server: FPBX-2.10.1(10.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:875@195.191.226.132:5060>
Content-Type: application/sdp
Content-Length: 337

v=0
o=root 365350069 365350069 IN IP4 195.191.226.132
s=Asterisk PBX 10.0.0

c=IN IP4 195.191.226.132
t=0 0
m=audio 11932 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:194.0.88.214:1250 --->
ACK sip:875@195.191.226.132:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK1681157186;rport
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
To: <sip:875@freepbx.edi.su>;tag=as779d8908
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 21 ACK
Contact: <sip:855@192.168.0.35:6892>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:855@192.168.0.35:6892> for address/port to send to
set_destination: set destination to 192.168.0.35:6892
Reliably Transmitting (NAT) to 194.0.88.214:1250:
INVITE sip:855@192.168.0.35:6892 SIP/2.0
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK2583afb9;rport
Max-Forwards: 70
From: <sip:875@freepbx.edi.su>;tag=as779d8908
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
Contact: <sip:875@195.191.226.132:5060>
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(10.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 365350069 365350070 IN IP4 195.191.226.132
s=Asterisk PBX 10.0.0
c=IN IP4 195.191.226.132
t=0 0
m=image 4355 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC

---

---

<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK2583afb9;rport=5060
From: <sip:875@freepbx.edi.su>;tag=as779d8908
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 102 INVITE
Contact: <sip:855@192.168.0.35:6892>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

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

<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK2583afb9;rport=5060
From: <sip:875@freepbx.edi.su>;tag=as779d8908
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 102 INVITE
Contact: <sip:855@192.168.0.35:6892>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 262

v=0
o=855 8000 8001 IN IP4 192.168.0.35
s=SIP Call
c=IN IP4 192.168.0.35
t=0 0
m=image 8577 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<------------->

<------------->
--- (12 headers 12 lines) ---
Got T.38 offer in SDP in dialog 622459119-6892-3@BJC.BGI.A.DF
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (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:855@192.168.0.35:6892> for address/port to send to
set_destination: set destination to 192.168.0.35:6892
Transmitting (NAT) to 194.0.88.214:1250:
ACK sip:855@192.168.0.35:6892 SIP/2.0
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK5c2eb79d;rport
Max-Forwards: 70
From: <sip:875@freepbx.edi.su>;tag=as779d8908
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
Contact: <sip:875@195.191.226.132:5060>
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 102 ACK
User-Agent: FPBX-2.10.1(10.0.0)
Content-Length: 0


---
-- Channel 'SIP/855-00000000' FAX session '0' started
> Saved useragent "Linksys/SPA8000-6.1.10(001)" for peer 334

<--- SIP read from UDP:194.0.88.214:1250 --->
REGISTER sip:freepbx.edi.su SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK589258724;rport
Route: <sip:freepbx.edi.su:5060;lr>
From: "FAX_test" <sip:855@freepbx.edi.su>;tag=1725882244
To: <sip:855@freepbx.edi.su>
Call-ID: 853023192-6892-1
CSeq: 2181 REGISTER
Contact: <sip:855@192.168.0.35:6892>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B823AB093>"
Authorization: Digest username="855", realm="asterisk", nonce="691a262a", uri="sip:freepbx.edi.su", response="e0e01cc42c30cdf5a3dfbe71b7a95638", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Supported: path
Expires: 300
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 194.0.88.214:1250 (NAT)

<--- Transmitting (NAT) to 194.0.88.214:1250 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK589258724;received=194.0.88.214;rport=1250
From: "FAX_test" <sip:855@freepbx.edi.su>;tag=1725882244
To: <sip:855@freepbx.edi.su>;tag=as429c81af
Call-ID: 853023192-6892-1
CSeq: 2181 REGISTER
Server: FPBX-2.10.1(10.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e941d71"
Content-Length: 0


CSeq: 2181 REGISTER
Server: FPBX-2.10.1(10.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e941d71"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '853023192-6892-1' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:194.0.88.214:1250 --->
REGISTER sip:freepbx.edi.su SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK248665393;rport
Route: <sip:freepbx.edi.su:5060;lr>
From: "FAX_test" <sip:855@freepbx.edi.su>;tag=1725882244
To: <sip:855@freepbx.edi.su>
Call-ID: 853023192-6892-1
CSeq: 2182 REGISTER
Contact: <sip:855@192.168.0.35:6892>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B823AB093>"
Authorization: Digest username="855", realm="asterisk", nonce="1e941d71", uri="sip:freepbx.edi.su", response="5dc813ee2b0fe935022996b68c22e731", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Supported: path
Expires: 300
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 194.0.88.214:1250 (NAT)
Reliably Transmitting (NAT) to 194.0.88.214:1250:
OPTIONS sip:855@192.168.0.35:6892 SIP/2.0
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK0991e62a;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@195.191.226.132>;tag=as447d5b1a
To: <sip:855@192.168.0.35:6892>
Contact: <sip:Unknown@195.191.226.132:5060>
Call-ID: 53ca480f6beed1375b41ad0e5c98b8e4@195.191.226.132:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(10.0.0)
Date: Wed, 10 Oct 2012 15:20:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK248665393;received=194.0.88.214;rport=1250
From: "FAX_test" <sip:855@freepbx.edi.su>;tag=1725882244
To: <sip:855@freepbx.edi.su>;tag=as429c81af
Call-ID: 853023192-6892-1
CSeq: 2182 REGISTER
Server: FPBX-2.10.1(10.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:855@192.168.0.35:6892>;expires=300
Date: Wed, 10 Oct 2012 15:20:31 GMT
Content-Length: 0


---
Scheduling destruction of SIP dialog '853023192-6892-1' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK0991e62a;rport=5060
From: "Unknown" <sip:Unknown@195.191.226.132>;tag=as447d5b1a
To: <sip:855@192.168.0.35:6892>;tag=1879251389
Call-ID: 53ca480f6beed1375b41ad0e5c98b8e4@195.191.226.132:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3b79f2ef65ca169c778f37fc0ddd907c@195.191.226.132:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 194.0.88.214:1250:
NOTIFY sip:855@192.168.0.35:6892 SIP/2.0
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK4877502d;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@195.191.226.132>;tag=as51345748
To: <sip:855@192.168.0.35:6892>
Contact: <sip:Unknown@195.191.226.132:5060>
Call-ID: 3b79f2ef65ca169c778f37fc0ddd907c@195.191.226.132:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.10.1(10.0.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 95

Messages-Waiting: no
Message-Account: sip:*97@195.191.226.132:5060
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog '853023192-6892-1' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK0991e62a;rport=5060
From: "Unknown" <sip:Unknown@195.191.226.132>;tag=as447d5b1a
To: <sip:855@192.168.0.35:6892>;tag=1879251389
Call-ID: 53ca480f6beed1375b41ad0e5c98b8e4@195.191.226.132:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '53ca480f6beed1375b41ad0e5c98b8e4@195.191.226.132:5060' Method: OPTIONS

<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK4877502d;rport=5060
From: "Unknown" <sip:Unknown@195.191.226.132>;tag=as51345748
To: <sip:855@192.168.0.35:6892>;tag=1708348571
Call-ID: 3b79f2ef65ca169c778f37fc0ddd907c@195.191.226.132:5060
CSeq: 102 NOTIFY
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

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

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '53ca480f6beed1375b41ad0e5c98b8e4@195.191.226.132:5060' Method: OPTIONS

<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK4877502d;rport=5060
From: "Unknown" <sip:Unknown@195.191.226.132>;tag=as51345748
To: <sip:855@192.168.0.35:6892>;tag=1708348571
Call-ID: 3b79f2ef65ca169c778f37fc0ddd907c@195.191.226.132:5060
CSeq: 102 NOTIFY
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3b79f2ef65ca169c778f37fc0ddd907c@195.191.226.132:5060' Method: NOTIFY
-- Registered SIP '863' at 176.36.121.126:11871
> Saved useragent "X-Lite release 1103k stamp 53621" for peer 863
-- Got SIP response 603 "Decline" back from 176.36.121.126:11871

<--- SIP read from UDP:194.0.88.214:1250 --->
BYE sip:875@195.191.226.132:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK1887311370;rport
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
To: <sip:875@freepbx.edi.su>;tag=as779d8908
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 22 BYE
Contact: <sip:855@192.168.0.35:6892>
Max-Forwards: 70

Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '53ca480f6beed1375b41ad0e5c98b8e4@195.191.226.132:5060' Method: OPTIONS

<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK4877502d;rport=5060
From: "Unknown" <sip:Unknown@195.191.226.132>;tag=as51345748
To: <sip:855@192.168.0.35:6892>;tag=1708348571
Call-ID: 3b79f2ef65ca169c778f37fc0ddd907c@195.191.226.132:5060
CSeq: 102 NOTIFY
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3b79f2ef65ca169c778f37fc0ddd907c@195.191.226.132:5060' Method: NOTIFY
-- Registered SIP '863' at 176.36.121.126:11871
> Saved useragent "X-Lite release 1103k stamp 53621" for peer 863
-- Got SIP response 603 "Decline" back from 176.36.121.126:11871

<--- SIP read from UDP:194.0.88.214:1250 --->
BYE sip:875@195.191.226.132:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK1887311370;rport
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
To: <sip:875@freepbx.edi.su>;tag=as779d8908
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 22 BYE
Contact: <sip:855@192.168.0.35:6892>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 194.0.88.214:1250 (NAT)
Scheduling destruction of SIP dialog '622459119-6892-3@BJC.BGI.A.DF' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.35:6892;branch=z9hG4bK1887311370;received=194.0.88.214;rport=1250
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1724600832
To: <sip:875@freepbx.edi.su>;tag=as779d8908
Call-ID: 622459119-6892-3@BJC.BGI.A.DF
CSeq: 22 BYE
Server: FPBX-2.10.1(10.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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




[2012-10-10 18:21:01] WARNING[32540]: chan_sip.c:3892 __sip_autodestruct: Autodestruct on dialog '622459119-6892-3@BJC.BGI.A.DF' with owner in place (Method: BYE)
[2012-10-10 18:21:01] WARNING[32540]: chan_sip.c:3892 __sip_autodestruct: Autodestruct on dialog '622459119-6892-3@BJC.BGI.A.DF' with owner in place (Method: BYE)
[2012-10-10 18:21:01] WARNING[32540]: chan_sip.c:3892 __sip_autodestruct: Autodestruct on dialog '622459119-6892-3@BJC.BGI.A.DF' with owner in place (Method: BYE)
[2012-10-10 18:21:01] WARNING[32540]: chan_sip.c:3892 __sip_autodestruct: Autodestruct on dialog '622459119-6892-3@BJC.BGI.A.DF' with owner in place (Method: BYE)
[2012-10-10 18:21:01] WARNING[32540]: chan_sip.c:3892 __sip_autodestruct: Autodestruct on dialog '622459119-6892-3@BJC.BGI.A.DF' with owner in place (Method: BYE)
[2012-10-10 18:21:01] WARNING[32540]: chan_sip.c:3892 __sip_autodestruct: Autodestruct on dialog '622459119-6892-3@BJC.BGI.A.DF' with owner in place (Method: BYE)
[2012-10-10 18:21:01] WARNING[32540]: chan_sip.c:3892 __sip_autodestruct: Autodestruct on dialog '622459119-6892-3@BJC.BGI.A.DF' with owner in place (Method: BYE)
[2012-10-10 18:21:01] WARNING[32540]: chan_sip.c:3892 __sip_autodestruct: Autodestruct on dialog '622459119-6892-3@BJC.BGI.A.DF' with owner in place (Method: BYE)
[2012-10-10 18:21:01] WARNING[32540]: chan_sip.c:3892 __sip_autodestruct: Autodestruct on dialog '622459119-6892-3@BJC.BGI.A.DF' with owner in place (Method: BYE)
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:13076 sip_reregister: -- Re-registration for 0034665223@212.53.40.40
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:13076 sip_reregister: -- Re-registration for 0034665223@212.53.40.40
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:13076 sip_reregister: -- Re-registration for 0034665223@212.53.40.40
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:13076 sip_reregister: -- Re-registration for 0034665223@212.53.40.40
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:13076 sip_reregister: -- Re-registration for 0034665223@212.53.40.40
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:13076 sip_reregister: -- Re-registration for 0034665223@212.53.40.40
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:13076 sip_reregister: -- Re-registration for 0034665223@212.53.40.40
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:13076 sip_reregister: -- Re-registration for 0034665223@212.53.40.40
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:13076 sip_reregister: -- Re-registration for 0034665223@212.53.40.40
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:20895 handle_response_register: Outbound Registration: Expiry for 212.53.40.40 is 116 sec (Scheduling reregistration in 101 s)
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:20895 handle_response_register: Outbound Registration: Expiry for 212.53.40.40 is 116 sec (Scheduling reregistration in 101 s)
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:20895 handle_response_register: Outbound Registration: Expiry for 212.53.40.40 is 116 sec (Scheduling reregistration in 101 s)
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:20895 handle_response_register: Outbound Registration: Expiry for 212.53.40.40 is 116 sec (Scheduling reregistration in 101 s)
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:20895 handle_response_register: Outbound Registration: Expiry for 212.53.40.40 is 116 sec (Scheduling reregistration in 101 s)
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:20895 handle_response_register: Outbound Registration: Expiry for 212.53.40.40 is 116 sec (Scheduling reregistration in 101 s)
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:20895 handle_response_register: Outbound Registration: Expiry for 212.53.40.40 is 116 sec (Scheduling reregistration in 101 s)
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:20895 handle_response_register: Outbound Registration: Expiry for 212.53.40.40 is 116 sec (Scheduling reregistration in 101 s)
[2012-10-10 18:21:03] NOTICE[32540]: chan_sip.c:20895 handle_response_register: Outbound Registration: Expiry for 212.53.40.40 is 116 sec (Scheduling reregistration in 101 s)
Really destroying SIP dialog '853023192-6892-1' Method: REGISTER
[2012-10-10 18:21:04] ERROR[313]: res_fax.c:1476 generic_fax_exec: channel 'SIP/855-00000000' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT)
[2012-10-10 18:21:04] ERROR[313]: res_fax.c:1476 generic_fax_exec: channel 'SIP/855-00000000' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT)
[2012-10-10 18:21:04] ERROR[313]: res_fax.c:1476 generic_fax_exec: channel 'SIP/855-00000000' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT)
[2012-10-10 18:21:04] ERROR[313]: res_fax.c:1476 generic_fax_exec: channel 'SIP/855-00000000' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT)
[2012-10-10 18:21:04] ERROR[313]: res_fax.c:1476 generic_fax_exec: channel 'SIP/855-00000000' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT)
[2012-10-10 18:21:04] ERROR[313]: res_fax.c:1476 generic_fax_exec: channel 'SIP/855-00000000' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT)
[2012-10-10 18:21:04] ERROR[313]: res_fax.c:1476 generic_fax_exec: channel 'SIP/855-00000000' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT)
[2012-10-10 18:21:04] ERROR[313]: res_fax.c:1476 generic_fax_exec: channel 'SIP/855-00000000' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT)
[2012-10-10 18:21:04] ERROR[313]: res_fax.c:1476 generic_fax_exec: channel 'SIP/855-00000000' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT)
== Spawn extension (infax, receive, 19) exited non-zero on 'SIP/855-00000000'
-- Executing [h@infax:1] NoOp("SIP/855-00000000", "FAXOPT(ecm) : yes") in new stack
-- Executing [h@infax:2] NoOp("SIP/855-00000000", "FAXOPT(filename) : /var/www/html/in_fax/fax-855-1.tif") in new stack
-- Executing [h@infax:3] NoOp("SIP/855-00000000", "FAXOPT(headerinfo) : MY FAXBACK RX") in new stack
-- Executing [h@infax:4] NoOp("SIP/855-00000000", "FAXOPT(localstationid) : 875") in new stack
-- Executing [h@infax:5] NoOp("SIP/855-00000000", "FAXOPT(maxrate) : 14400") in new stack
-- Executing [h@infax:6] NoOp("SIP/855-00000000", "FAXOPT(minrate) : 2400") in new stack
-- Executing [h@infax:7] NoOp("SIP/855-00000000", "FAXOPT(pages) : 0") in new stack
-- Executing [h@infax:8] NoOp("SIP/855-00000000", "FAXOPT(rate) : ") in new stack
-- Executing [h@infax:9] NoOp("SIP/855-00000000", "FAXOPT(remotestationid) : ") in new stack
-- Executing [h@infax:10] NoOp("SIP/855-00000000", "FAXOPT(resolution) : ") in new stack
-- Executing [h@infax:11] NoOp("SIP/855-00000000", "FAXOPT(status) : FAILED") in new stack
-- Executing [h@infax:12] NoOp("SIP/855-00000000", "FAXOPT(statusstr) : remote channel hungup") in new stack
-- Executing [h@infax:13] NoOp("SIP/855-00000000", "FAXOPT(error) : HANGUP") in new stack
внешние провайдеры работают как с т38 так и g711.
Подскажите в чем может быть причина.
Вернуться к началу
Morya4ok-cr
Сообщения: 6
Зарегистрирован: 10 окт 2012, 19:04

Re: Free Fax for Asterisk неработает ReciveFax

Сообщение Morya4ok-cr »

вот еще попытка после которой сервер вылетел
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK66342508;rport=5060
From: <sip:875@freepbx.edi.su>;tag=as58c24e6f
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=635617507
Call-ID: 630778086-6892-4@BJC.BGI.A.DF
CSeq: 102 INVITE
Contact: <sip:855@192.168.0.35:6892>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 262

v=0
o=855 8000 8001 IN IP4 192.168.0.35
s=SIP Call
c=IN IP4 192.168.0.35
t=0 0
m=image 8577 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<------------->
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c:
<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK66342508;rport=5060
From: <sip:875@freepbx.edi.su>;tag=as58c24e6f
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=635617507
Call-ID: 630778086-6892-4@BJC.BGI.A.DF
CSeq: 102 INVITE
Contact: <sip:855@192.168.0.35:6892>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 262

v=0
o=855 8000 8001 IN IP4 192.168.0.35
s=SIP Call
s=SIP Call
c=IN IP4 192.168.0.35
t=0 0
m=image 8577 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<------------->
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c:
<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK66342508;rport=5060
From: <sip:875@freepbx.edi.su>;tag=as58c24e6f
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=635617507
Call-ID: 630778086-6892-4@BJC.BGI.A.DF
CSeq: 102 INVITE
Contact: <sip:855@192.168.0.35:6892>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 262

v=0
o=855 8000 8001 IN IP4 192.168.0.35
s=SIP Call
c=IN IP4 192.168.0.35
t=0 0
m=image 8577 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<------------->
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c:
<--- SIP read from UDP:194.0.88.214:1250 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK66342508;rport=5060
From: <sip:875@freepbx.edi.su>;tag=as58c24e6f
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=635617507
Call-ID: 630778086-6892-4@BJC.BGI.A.DF
CSeq: 102 INVITE
Contact: <sip:855@192.168.0.35:6892>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXW-4024 V0.4A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 262

v=0
o=855 8000 8001 IN IP4 192.168.0.35
s=SIP Call
c=IN IP4 192.168.0.35
t=0 0
m=image 8577 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<------------->
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: --- (12 headers 12 lines) ---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: --- (12 headers 12 lines) ---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: --- (12 headers 12 lines) ---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: --- (12 headers 12 lines) ---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: --- (12 headers 12 lines) ---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: --- (12 headers 12 lines) ---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: --- (12 headers 12 lines) ---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: --- (12 headers 12 lines) ---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: --- (12 headers 12 lines) ---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 offer in SDP in dialog 630778086-6892-4@BJC.BGI.A.DF
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 offer in SDP in dialog 630778086-6892-4@BJC.BGI.A.DF
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 offer in SDP in dialog 630778086-6892-4@BJC.BGI.A.DF
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 offer in SDP in dialog 630778086-6892-4@BJC.BGI.A.DF
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 offer in SDP in dialog 630778086-6892-4@BJC.BGI.A.DF
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 offer in SDP in dialog 630778086-6892-4@BJC.BGI.A.DF
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 offer in SDP in dialog 630778086-6892-4@BJC.BGI.A.DF
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 offer in SDP in dialog 630778086-6892-4@BJC.BGI.A.DF
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 offer in SDP in dialog 630778086-6892-4@BJC.BGI.A.DF
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: Parsing <sip:855@192.168.0.35:6892> for address/port to send to
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: Parsing <sip:855@192.168.0.35:6892> for address/port to send to
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: Parsing <sip:855@192.168.0.35:6892> for address/port to send to
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: Parsing <sip:855@192.168.0.35:6892> for address/port to send to
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: set destination to 192.168.0.35:6892
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: set destination to 192.168.0.35:6892
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: set destination to 192.168.0.35:6892
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: set destination to 192.168.0.35:6892
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: set destination to 192.168.0.35:6892
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: set destination to 192.168.0.35:6892
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: set destination to 192.168.0.35:6892
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: set_destination: set destination to 192.168.0.35:6892
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Transmitting (NAT) to 194.0.88.214:1250:
ACK sip:855@192.168.0.35:6892 SIP/2.0
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK5134edf0;rport
Max-Forwards: 70
From: <sip:875@freepbx.edi.su>;tag=as58c24e6f
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=635617507
Contact: <sip:875@195.191.226.132:5060>
Call-ID: 630778086-6892-4@BJC.BGI.A.DF
CSeq: 102 ACK
User-Agent: FPBX-2.10.1(10.0.0)
Content-Length: 0


---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Transmitting (NAT) to 194.0.88.214:1250:
ACK sip:855@192.168.0.35:6892 SIP/2.0
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK5134edf0;rport
Max-Forwards: 70
From: <sip:875@freepbx.edi.su>;tag=as58c24e6f
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=635617507
Contact: <sip:875@195.191.226.132:5060>
Call-ID: 630778086-6892-4@BJC.BGI.A.DF
CSeq: 102 ACK
User-Agent: FPBX-2.10.1(10.0.0)
Content-Length: 0


---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Transmitting (NAT) to 194.0.88.214:1250:
ACK sip:855@192.168.0.35:6892 SIP/2.0
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK5134edf0;rport
Max-Forwards: 70
From: <sip:875@freepbx.edi.su>;tag=as58c24e6f
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=635617507
Contact: <sip:875@195.191.226.132:5060>
Call-ID: 630778086-6892-4@BJC.BGI.A.DF
CSeq: 102 ACK
User-Agent: FPBX-2.10.1(10.0.0)
Content-Length: 0


---
[2012-10-10 18:46:22] VERBOSE[32540] chan_sip.c: Transmitting (NAT) to 194.0.88.214:1250:
ACK sip:855@192.168.0.35:6892 SIP/2.0
Via: SIP/2.0/UDP 195.191.226.132:5060;branch=z9hG4bK5134edf0;rport
Max-Forwards: 70
From: <sip:875@freepbx.edi.su>;tag=as58c24e6f
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=635617507
Contact: <sip:875@195.191.226.132:5060>
Call-ID: 630778086-6892-4@BJC.BGI.A.DF
CSeq: 102 ACK
User-Agent: FPBX-2.10.1(10.0.0)
Content-Length: 0
Content-Length: 0


---
[2012-10-10 18:46:22] VERBOSE[2573] res_fax_digium.c: -- Channel 'SIP/855-00000008' FAX session '1' started
[2012-10-10 18:46:22] VERBOSE[2573] res_fax_digium.c: -- Channel 'SIP/855-00000008' FAX session '1' started
[2012-10-10 18:46:22] VERBOSE[2573] res_fax_digium.c: -- Channel 'SIP/855-00000008' FAX session '1' started
[2012-10-10 18:46:22] VERBOSE[2573] res_fax_digium.c: -- Channel 'SIP/855-00000008' FAX session '1' started
[2012-10-10 18:46:22] VERBOSE[2573] res_fax_digium.c: -- Channel 'SIP/855-00000008' FAX session '1' started
[2012-10-10 18:46:22] VERBOSE[2573] res_fax_digium.c: -- Channel 'SIP/855-00000008' FAX session '1' started
[2012-10-10 18:46:22] VERBOSE[2573] res_fax_digium.c: -- Channel 'SIP/855-00000008' FAX session '1' started
[2012-10-10 18:46:22] VERBOSE[2573] res_fax_digium.c: -- Channel 'SIP/855-00000008' FAX session '1' started
[2012-10-10 18:46:22] VERBOSE[2573] res_fax_digium.c: -- Channel 'SIP/855-00000008' FAX session '1' started
[2012-10-10 18:46:22] WARNING[2573] udptl.c: UDPTL (SIP/855-00000008): UDPTL can only send T.38 data.
[2012-10-10 18:49:07] NOTICE[2831] loader.c: 2 modules will be loaded.
[2012-10-10 18:49:07] NOTICE[2831] loader.c: 2 modules will be loaded.
[2012-10-10 18:49:07] NOTICE[2831] loader.c: 2 modules will be loaded.
[2012-10-10 18:49:07] NOTICE[2831] loader.c: 2 modules will be loaded.
[2012-10-10 18:49:07] NOTICE[2831] loader.c: 2 modules will be loaded.
[2012-10-10 18:49:07] NOTICE[2831] loader.c: 2 modules will be loaded.
[2012-10-10 18:49:07] NOTICE[2831] loader.c: 2 modules will be loaded.
[2012-10-10 18:49:07] NOTICE[2831] loader.c: 2 modules will be loaded.
[2012-10-10 18:49:07] NOTICE[2831] loader.c: 2 modules will be loaded.
[2012-10-10 18:49:07] WARNING[2831] pbx.c: Maximum PBX stack exceeded
[2012-10-10 18:49:07] WARNING[2831] pbx.c: Maximum PBX stack exceeded
[2012-10-10 18:49:07] WARNING[2831] pbx.c: Maximum PBX stack exceeded
[2012-10-10 18:49:07] WARNING[2831] pbx.c: Maximum PBX stack exceeded
[2012-10-10 18:49:07] WARNING[2831] pbx.c: Maximum PBX stack exceeded
[2012-10-10 18:49:07] WARNING[2831] pbx.c: Maximum PBX stack exceeded
[2012-10-10 18:49:07] WARNING[2831] pbx.c: Maximum PBX stack exceeded
[2012-10-10 18:49:07] WARNING[2831] pbx.c: Maximum PBX stack exceeded
[2012-10-10 18:49:07] WARNING[2831] pbx.c: Maximum PBX stack exceeded
в sip.conf

Код: Выделить всё

disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
callevents=no
language=ru
bindaddr=0.0.0.0
jbenable=no
defaultexpiry=120
allowguest=yes
srvlookup=no
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
rtpkeepalive=0
g726nonstandard=no
t38pt_udptl=yes
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
checkmwi=10
notifyhold=yes
notifyringing=yes
nat=yes

выдало ошибку при отправке

Код: Выделить всё

[2012-10-10 18:59:43] ERROR[3713]: res_fax.c:1476 generic_fax_exec: channel 'SIP/855-00000000' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT)
Вернуться к началу
Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: Free Fax for Asterisk неработает ReciveFax

Сообщение Vlad1983 »

http://ictfax.org/content/asterisk-18-installation
ЛС: @rostel
Вернуться к началу
Morya4ok-cr
Сообщения: 6
Зарегистрирован: 10 окт 2012, 19:04

Re: Free Fax for Asterisk неработает ReciveFax

Сообщение Morya4ok-cr »

Пасиба, уже думал перекомпилить Asterisk и поставить SpanDSP но пока еще не решился
Вернуться к началу
awsswa
Сообщения: 2390
Зарегистрирован: 09 июн 2012, 10:52
Откуда: Россия, Пермь skype: yarick_perm

Re: Free Fax for Asterisk неработает ReciveFax

Сообщение awsswa »

Тут и пошаговая инструкция есть
http://www.pbxinaflash.com/community/in ... 6-2.14209/

а что только модуль добавить - после make, не делайте make install - просто скопируйте ручками собранный res_fax_spandsp
платный суппорт по мере возможностей
Вернуться к началу
Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: Free Fax for Asterisk неработает ReciveFax

Сообщение Vlad1983 »

awsswa писал(а):а что только модуль добавить - после make, не делайте make install - просто скопируйте ручками собранный res_fax_spandsp
самый верный путь поиметь гемор
ЛС: @rostel
Вернуться к началу
awsswa
Сообщения: 2390
Зарегистрирован: 09 июн 2012, 10:52
Откуда: Россия, Пермь skype: yarick_perm

Re: Free Fax for Asterisk неработает ReciveFax

Сообщение awsswa »

само сабой - это путь для настоящих чингачгуков.
платный суппорт по мере возможностей
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Morya4ok-cr
Сообщения: 6
Зарегистрирован: 10 окт 2012, 19:04

Re: Free Fax for Asterisk неработает ReciveFax

Сообщение Morya4ok-cr »

Пошел путем "настаящих ченгачкуков" перекомпилил без "make install" и ручками все по копировал, вроде не че не поломал)
потестить факс не успел, но модуль появился

Код: Выделить всё

freepbx*CLI> module show like res_fax_spandsp.so
Module                         Description                              Use Count
res_fax_spandsp.so             Spandsp G.711 and T.38 FAX Technologies  0
завтра буду пробывать игратся с базой, чтобы на нужные адреса факсы уходили
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ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Free Fax for Asterisk неработает ReciveFax

Сообщение ded »

Вы ещё на послезавтра ваше расписание задач нам сообщите. И на всю неделю. Сделали - хорошо, не доделали - плохо, промежуточные результаты - только в личном блоге.
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