asterisk.ru

Русский форум Asterisk сообщества
https://forum.asterisk.ru/

Не понятно как полностью отключить T38

https://forum.asterisk.ru/viewtopic.php?t=4253

Страница 1 из 2

Не понятно как полностью отключить T38

Добавлено: 02 апр 2014, 12:59
SilentNess
у провайдера отключен t38, факсы пытаемся принять по g711, при снятии tcpdump видно что наша сторона сначала согласна на g711 а потом шлет invate по t38, скрин прилагается, подскажите пожалуйста в чем может быть проблема

Re: Не понятно как полностью отключить T38

Добавлено: 02 апр 2014, 13:09
ded
Возможно у вас стоит где-то автодетект факса? Тогда если в голосовой сессии по G711 Астериск слышит писк факса на удалённой стороне - он переключается в факсовый режим и шлёт реинвайт Т38? Это можно видеть в логе консоли, особенно если включить sip debug.

Re: Не понятно как полностью отключить T38

Добавлено: 02 апр 2014, 13:17
SilentNess
faxdetect=no стоит в sip.conf и на транке с провайдером, изменений нет

Re: Не понятно как полностью отключить T38

Добавлено: 02 апр 2014, 13:21
ded
sip set debug ip ip_addr_operator
и смотрим в CLI при посылке факса.

Re: Не понятно как полностью отключить T38

Добавлено: 02 апр 2014, 13:36
SilentNess

Код: Выделить всё

== Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
set_destination: Parsing <sip:8XXXX203040@20.56.240.19:5060;user=phone> for address/port to send to
set_destination: set destination to 20.56.240.19:5060
Reliably Transmitting (NAT) to 20.56.240.19:5060:
INVITE sip:8XXXX203040@20.56.240.19:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK2c4b104f;rport
Max-Forwards: 70
From: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as64b9844b
To: <sip:8XXXX203040@20.56.240.19>;tag=as7s8qad-CC-23
Contact: <sip:XXXX980660@20.56.1.14:5060>
Call-ID: SBC2vama7zq8s733023m35arrt0v2a88zap@SoftX3000
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.20.2)
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: 260

v=0
o=root 908701294 908701295 IN IP4 20.56.1.14
s=Asterisk PBX 1.8.20.2
c=IN IP4 20.56.1.14
t=0 0
m=image 20016 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC

---

<--- SIP read from UDP:20.56.240.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK2c4b104f;rport=5060
Call-ID: SBC2vama7zq8s733023m35arrt0v2a88zap@SoftX3000
From: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as64b9844b
To: <sip:8XXXX203040@20.56.240.19>;tag=as7s8qad-CC-23
CSeq: 102 INVITE
Content-Length: 0

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


<--- SIP read from UDP:20.56.240.19:5060 --->
BYE sip:XXXX980660@20.56.1.14:5060 SIP/2.0
Via: SIP/2.0/UDP 20.56.240.19:5060;branch=z9hG4bKco8xkveouu88ji7enchlih7ob
Call-ID: SBC2vama7zq8s733023m35arrt0v2a88zap@SoftX3000
From: <sip:8XXXX203040@20.56.240.19>;tag=as7s8qad-CC-23
To: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as64b9844b
CSeq: 2 BYE
Max-Forwards: 70
Reason: Q.850;cause=57;text="bearer capability not authorized"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 20.56.240.19:5060 (NAT)
Scheduling destruction of SIP dialog 'SBC2vama7zq8s733023m35arrt0v2a88zap@SoftX3000' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 20.56.240.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 20.56.240.19:5060;branch=z9hG4bKco8xkveouu88ji7enchlih7ob;received=20.56.240.19;rport=5060
From: <sip:8XXXX203040@20.56.240.19>;tag=as7s8qad-CC-23
To: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as64b9844b
Call-ID: SBC2vama7zq8s733023m35arrt0v2a88zap@SoftX3000
CSeq: 2 BYE
Server: FPBX-2.9.0(1.8.20.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

Re: Не понятно как полностью отключить T38

Добавлено: 02 апр 2014, 13:39
SilentNess
я так понимаю вот это

Re: Не понятно как полностью отключить T38

Добавлено: 02 апр 2014, 13:42
ded
Там у вас ивайт уже содержит
m=image 20016 udptl t38
то есть это уже после реинвайта.

Непонятно откуда и куда пересылается,

Re: Не понятно как полностью отключить T38

Добавлено: 02 апр 2014, 13:53
SilentNess

Код: Выделить всё

<--- SIP read from UDP:20.56.240.19:5060 --->
INVITE sip:XXXX980660@20.56.1.14:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 20.56.240.19:5060;branch=z9hG4bK7nknycvjujlxcuhkecn7ehcil
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
From: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
To: <sip:XXXX980660@20.56.1.14;user=phone>
CSeq: 1 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Max-Forwards: 70
Supported: 100rel
User-Agent: Huawei SoftX3000 V300R010
Contact: <sip:8XXXX203040@20.56.240.19:5060;user=phone>
Content-Length: 388
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 285095 285095 IN IP4 20.56.240.19
s=Sip Call
c=IN IP4 20.56.240.19
t=0 0
m=audio 37402 RTP/AVP 8 18 4 2 98 99 102 97
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:102 G726-24/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes
<------------->
--- (13 headers 16 lines) ---
Sending to 20.56.240.19:5060 (NAT)
Using INVITE request as basis request - SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
Found peer '203359' for '8XXXX203040' from 20.56.240.19:5060
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 102
Found RTP audio format 97
Found audio description format PCMA for ID 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 unknown media description format G726-40 for ID 98
Found audio description format G726-32 for ID 99
Found unknown media description format G726-24 for ID 102
Found audio description format telephone-event for ID 97
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x909 (g723|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 20.56.240.19:37402
Looking for XXXX980660 in from-trunk (domain 20.56.1.14)
list_route: hop: <sip:8XXXX203040@20.56.240.19:5060;user=phone>

<--- Transmitting (NAT) to 20.56.240.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 20.56.240.19:5060;branch=z9hG4bK7nknycvjujlxcuhkecn7ehcil;received=20.56.240.19;rport=5060
From: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
To: <sip:XXXX980660@20.56.1.14;user=phone>
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.20.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:XXXX980660@20.56.1.14:5060>
Content-Length: 0


<------------>
Audio is at 39824
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (NAT) to 20.56.240.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 20.56.240.19:5060;branch=z9hG4bK7nknycvjujlxcuhkecn7ehcil;received=20.56.240.19;rport=5060
From: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
To: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as54453bb9
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.20.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:XXXX980660@20.56.1.14:5060>
Content-Type: application/sdp
Content-Length: 175

v=0
o=root 532973087 532973087 IN IP4 20.56.1.14
s=Asterisk PBX 1.8.20.2
c=IN IP4 20.56.1.14
t=0 0
m=audio 39824 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (NAT) to 20.56.240.19:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 20.56.240.19:5060;branch=z9hG4bK7nknycvjujlxcuhkecn7ehcil;received=20.56.240.19;rport=5060
From: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
To: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as54453bb9
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.20.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:XXXX980660@20.56.1.14:5060>
Content-Type: application/sdp
Content-Length: 175

v=0
o=root 532973087 532973087 IN IP4 20.56.1.14
s=Asterisk PBX 1.8.20.2
c=IN IP4 20.56.1.14
t=0 0
m=audio 39824 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:20.56.240.19:5060 --->
ACK sip:XXXX980660@20.56.1.14:5060 SIP/2.0
Via: SIP/2.0/UDP 20.56.240.19:5060;branch=z9hG4bKblnoklbxueblyoyvjoe7uhnyc
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
From: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
To: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as54453bb9
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
[Apr  2 15:47:17] WARNING[19219]: sig_pri.c:6170 pri_dchannel: PRI Error on span 2: We think we're the CPE, but they think they're the CPE too.

[Apr  2 15:47:18] WARNING[19219]: sig_pri.c:6170 pri_dchannel: PRI Error on span 2: We think we're the CPE, but they think they're the CPE too.

Scheduling destruction of SIP dialog '0f92ac9712ae680b50f176cc1c954ffd@20.56.1.14:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 20.56.240.19:5060:
CANCEL sip:8XXXX139148@20.56.240.19 SIP/2.0
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK5ca76a45;rport
Max-Forwards: 70
From: "XXXX274329" <sip:XXXX274329@20.56.1.14>;tag=as4cad2fce
To: <sip:8XXXX139148@20.56.240.19>
Call-ID: 0f92ac9712ae680b50f176cc1c954ffd@20.56.1.14:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.20.2)
Content-Length: 0


---
[Apr  2 15:47:19] NOTICE[24100]: pbx_spool.c:372 attempt_thread: Call failed to go through, reason (3) Remote end Ringing
[Apr  2 15:47:19] NOTICE[24100]: pbx_spool.c:375 attempt_thread: Queued call to Local/1078XXXX139148@from-internal expired without completion after 0 attempts
Scheduling destruction of SIP dialog '0f92ac9712ae680b50f176cc1c954ffd@20.56.1.14:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '57c3b8837e2fd6b0719121dc473373e5@20.56.1.14:5060' in 6400 ms (Method: INVITE)
[Apr  2 15:47:19] NOTICE[24106]: pbx_spool.c:372 attempt_thread: Call failed to go through, reason (3) Remote end Ringing
[Apr  2 15:47:19] NOTICE[24106]: pbx_spool.c:375 attempt_thread: Queued call to Local/1078XXXX432767@from-internal expired without completion after 0 attempts
Reliably Transmitting (NAT) to 20.56.240.19:5060:
CANCEL sip:8XXXX432767@20.56.240.19 SIP/2.0
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK55f17529;rport
Max-Forwards: 70
From: "XXXX274329" <sip:XXXX274329@20.56.1.14>;tag=as2767d555
To: <sip:8XXXX432767@20.56.240.19>
Call-ID: 57c3b8837e2fd6b0719121dc473373e5@20.56.1.14:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.20.2)
Content-Length: 0


---
Scheduling destruction of SIP dialog '57c3b8837e2fd6b0719121dc473373e5@20.56.1.14:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '6be023a439a5fcec75152edd483e5f00@20.56.1.14:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 20.56.240.19:5060:
CANCEL sip:8XXXX921926@20.56.240.19 SIP/2.0
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK2b03111d;rport
Max-Forwards: 70
From: "XXXX274329" <sip:XXXX274329@20.56.1.14>;tag=as54d177d8
To: <sip:8XXXX921926@20.56.240.19>
Call-ID: 6be023a439a5fcec75152edd483e5f00@20.56.1.14:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.20.2)
Content-Length: 0


---
Scheduling destruction of SIP dialog '6be023a439a5fcec75152edd483e5f00@20.56.1.14:5060' in 6400 ms (Method: INVITE)
[Apr  2 15:47:19] NOTICE[24109]: pbx_spool.c:372 attempt_thread: Call failed to go through, reason (3) Remote end Ringing
[Apr  2 15:47:19] NOTICE[24109]: pbx_spool.c:375 attempt_thread: Queued call to Local/1078XXXX921926@from-internal expired without completion after 0 attempts

<--- SIP read from UDP:20.56.240.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK5ca76a45;rport=5060
Call-ID: 0f92ac9712ae680b50f176cc1c954ffd@20.56.1.14:5060
From: "XXXX274329"<sip:XXXX274329@20.56.1.14>;tag=as4cad2fce
To: <sip:8XXXX139148@20.56.240.19>;tag=oki7jjji
CSeq: 102 CANCEL
Content-Length: 0

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

<--- SIP read from UDP:20.56.240.19:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK5ca76a45;rport=5060
Call-ID: 0f92ac9712ae680b50f176cc1c954ffd@20.56.1.14:5060
From: "XXXX274329"<sip:XXXX274329@20.56.1.14>;tag=as4cad2fce
To: <sip:8XXXX139148@20.56.240.19>;tag=zdz00820-CC-24
CSeq: 102 INVITE
Warning: 399 SE2000 "SSF00156L00778[0000] Cancel received from network"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 20.56.240.19:5060:
ACK sip:8XXXX139148@20.56.240.19:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK5ca76a45;rport
Max-Forwards: 70
From: "XXXX274329" <sip:XXXX274329@20.56.1.14>;tag=as4cad2fce
To: <sip:8XXXX139148@20.56.240.19>;tag=zdz00820-CC-24
Contact: <sip:XXXX274329@20.56.1.14:5060>
Call-ID: 0f92ac9712ae680b50f176cc1c954ffd@20.56.1.14:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.20.2)
Content-Length: 0


---
Scheduling destruction of SIP dialog '0f92ac9712ae680b50f176cc1c954ffd@20.56.1.14:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:20.56.240.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK55f17529;rport=5060
Call-ID: 57c3b8837e2fd6b0719121dc473373e5@20.56.1.14:5060
From: "XXXX274329"<sip:XXXX274329@20.56.1.14>;tag=as2767d555
To: <sip:8XXXX432767@20.56.240.19>;tag=x8l7vhuj
CSeq: 102 CANCEL
Content-Length: 0

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

<--- SIP read from UDP:20.56.240.19:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK55f17529;rport=5060
Call-ID: 57c3b8837e2fd6b0719121dc473373e5@20.56.1.14:5060
From: "XXXX274329"<sip:XXXX274329@20.56.1.14>;tag=as2767d555
To: <sip:8XXXX432767@20.56.240.19>;tag=r85m583q-CC-23
CSeq: 102 INVITE
Warning: 399 SE2000 "SSF00156L00778[0000] Cancel received from network"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 20.56.240.19:5060:
ACK sip:8XXXX432767@20.56.240.19:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK55f17529;rport
Max-Forwards: 70
From: "XXXX274329" <sip:XXXX274329@20.56.1.14>;tag=as2767d555
To: <sip:8XXXX432767@20.56.240.19>;tag=r85m583q-CC-23
Contact: <sip:XXXX274329@20.56.1.14:5060>
Call-ID: 57c3b8837e2fd6b0719121dc473373e5@20.56.1.14:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.20.2)
Content-Length: 0


---
Scheduling destruction of SIP dialog '57c3b8837e2fd6b0719121dc473373e5@20.56.1.14:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:20.56.240.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK2b03111d;rport=5060
Call-ID: 6be023a439a5fcec75152edd483e5f00@20.56.1.14:5060
From: "XXXX274329"<sip:XXXX274329@20.56.1.14>;tag=as54d177d8
To: <sip:8XXXX921926@20.56.240.19>;tag=8vhvculb
CSeq: 102 CANCEL
Content-Length: 0

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

<--- SIP read from UDP:20.56.240.19:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK2b03111d;rport=5060
Call-ID: 6be023a439a5fcec75152edd483e5f00@20.56.1.14:5060
From: "XXXX274329"<sip:XXXX274329@20.56.1.14>;tag=as54d177d8
To: <sip:8XXXX921926@20.56.240.19>;tag=zr370z7v-CC-24
CSeq: 102 INVITE
Warning: 399 SE2000 "SSF00156L00778[0000] Cancel received from network"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 20.56.240.19:5060:
ACK sip:8XXXX921926@20.56.240.19:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK2b03111d;rport
Max-Forwards: 70
From: "XXXX274329" <sip:XXXX274329@20.56.1.14>;tag=as54d177d8
To: <sip:8XXXX921926@20.56.240.19>;tag=zr370z7v-CC-24
Contact: <sip:XXXX274329@20.56.1.14:5060>
Call-ID: 6be023a439a5fcec75152edd483e5f00@20.56.1.14:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.20.2)
Content-Length: 0


---
Scheduling destruction of SIP dialog '6be023a439a5fcec75152edd483e5f00@20.56.1.14:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:8XXXX203040@20.56.240.19:5060;user=phone> for address/port to send to
set_destination: set destination to 20.56.240.19:5060
Reliably Transmitting (NAT) to 20.56.240.19:5060:
INVITE sip:8XXXX203040@20.56.240.19:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK681be184;rport
Max-Forwards: 70
From: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as54453bb9
To: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
Contact: <sip:XXXX980660@20.56.1.14:5060>
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.20.2)
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: 260

v=0
o=root 532973087 532973088 IN IP4 20.56.1.14
s=Asterisk PBX 1.8.20.2
c=IN IP4 20.56.1.14
t=0 0
m=image 20014 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC

---

<--- SIP read from UDP:20.56.240.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK681be184;rport=5060
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
From: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as54453bb9
To: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:20.56.240.19:5060 --->
BYE sip:XXXX980660@20.56.1.14:5060 SIP/2.0
Via: SIP/2.0/UDP 20.56.240.19:5060;branch=z9hG4bKikuuok8uoocvc8y8bbyu8hnbn
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
From: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
To: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as54453bb9
CSeq: 2 BYE
Max-Forwards: 70
Reason: Q.850;cause=57;text="bearer capability not authorized"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 20.56.240.19:5060 (NAT)
Scheduling destruction of SIP dialog 'SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 20.56.240.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 20.56.240.19:5060;branch=z9hG4bKikuuok8uoocvc8y8bbyu8hnbn;received=20.56.240.19;rport=5060
From: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
To: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as54453bb9
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
CSeq: 2 BYE
Server: FPBX-2.9.0(1.8.20.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr  2 15:47:20] WARNING[24259]: res_fax.c:1654 receivefax_t38_init: error on 'SIP/203359-00000915' while waiting for T.38 negotiation.
[Apr  2 15:47:20] ERROR[24259]: res_fax.c:1915 receivefax_exec: error initializing channel 'SIP/203359-00000915' in T.38 mode

<--- SIP read from UDP:20.56.240.19:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK681be184;rport=5060
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
From: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as54453bb9
To: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
CSeq: 102 INVITE
Warning: 399 SE2000 "SSF00144L02533[5537] Request Terminated Due to Dialog Deletion"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:8XXXX203040@20.56.240.19:5060;user=phone> for address/port to send to
set_destination: set destination to 20.56.240.19:5060
Transmitting (NAT) to 20.56.240.19:5060:
ACK sip:8XXXX203040@20.56.240.19:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 20.56.1.14:5060;branch=z9hG4bK681be184;rport
Max-Forwards: 70
From: <sip:8XXXX203040@20.56.240.19>;tag=r33tv88v-CC-23
To: <sip:XXXX980660@20.56.1.14;user=phone>;tag=as54453bb9
Contact: <sip:XXXX980660@20.56.1.14:5060>
Call-ID: SBCt07383tmp8dd2ps28q5s0038mzqdprzs@SoftX3000
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.20.2)
Content-Length: 0

Re: Не понятно как полностью отключить T38

Добавлено: 02 апр 2014, 13:54
SilentNess
А если так?

Re: Не понятно как полностью отключить T38

Добавлено: 02 апр 2014, 14:03
ded
Вам бы самому это изучать, это же Ваши логи, верно?
Там уже видно навскидку, что ваш DAHDI настроен неправильно, Астериск говорит человеческим языком
[Apr 2 15:47:17] WARNING[19219]: sig_pri.c:6170 pri_dchannel: PRI Error on span 2: We think we're the CPE, but they think they're the CPE too.
то есть у вас указано, что ваш порт DAHDI является CPE, а должен быть NET относительно второй стороны по транку Е1. Наверное вторая сторона - офисная станция?
Ну и видны ответы SIP Cancel, после чего сопсно и сооружается реинвайт Т.38.
Почему?
Вам разбираться.
Часовой пояс: UTC+03:00
Страница 1 из 2

Создано на основе phpBB® Forum Software © phpBB Limited

Русская поддержка phpBB