есть схема: провайдер телефонии подключается на asterisk по sip, к asteriskу также подключается Addpac AP100, к адпаку в свою очередь на fxs порт подключается факс Panasonic KX-FT72. проблема с прохождениями факсов. факс из города нормально проходит, а вот факс с адпака в город не идет. конфиг такой:
провайдер:
[2112166]
type=peer
secret=******
username=2112166
host=91.144.140.130
fromuser=2112166
fromdomain=91.144.140.130
canreinvite=no
qualify=no
nat=yes
context=incoming
insecure=port,invite
allow=g729
dtmfmode=rfc2833
callgroup=1
pickupgroup=1
t38pt_udptl = yes,fec,maxdatagram=400

аддпак:
[406]
type=friend
host=dynamic
username=406
secret=********
nat=yes
canreinvite=no
context=incoming
callerid="Test" <406>
allow=g729
callgroup=1
pickupgroup=1
t38pt_udptl = yes,fec,maxdatagram=400

версия asterisk 1.6.2.6
в секции [general] тоже t38pt_udptl = yes

конфиг адпака:
voice service voip
fax protocol multi-session-t38 redundancy
fax rate 14400
h323 call start fast
h323 call tunnel enable
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 406
port 0/0
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
voice-class codec 0
no vad
dtmf-relay rtp-2833
!
!
!
!
!
!
! Gateway configuration.
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 2 g729

если нужны какие-то логи, напишите плиз, сниму.

Пир dial-peer voice 1000 voip
должен иметь строки
fax protocol ???
fax rate 14400

а
session target sip-server предполагает секцию sip-ua
типа
! SIP UA configuration.
!
sip-ua
user-register
sip-username AddPac-100
sip-password 1234567
sip-server 10.133.48.230
timeout tregtry 60
register e164
!

в секцию dial-peer voice 1000 voip добавил fax rate 14400 и пробовал добавлять fax protocol t38 redundancy 0 и fax protocol multi-session-t38 redundancy 0 - не помогло. а секция sip-ua она и до этого была, забыл ее написать, вот она:
sip-ua
sip-username 406
sip-password ********
sip-server 192.168.5.196
register e164

fax rate 14400 - это вряд ли, 9600 в самый раз.
- Лёд к месту ушиба приладывали?
- Да, не помогло!
- Зелёнкой мазали?
- Ой, чем только не мазали! И йодом, и горчицей, и кремом, и губной помадой...

Что мешает включать дебаг и анализировать причины? Сначала SIP дебаг, смотреть пакет invite sdp offer на предмет предложения T38, reinvite и ответов на invite.
Идеально - ничего этого не постить тут, а спокойно разобраться что к чему.

debug на asteriskе или на addpac?

Лично мне было бы интересно с двух сторон.

с адпака:
Received SIP PDU from ( 192.168.5.196:5060 )
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK19000f15a410;received=192.168.
5.152
From: <sip:406@192.168.5.196>;tag=19000f15a4
To: <sip:92614659@192.168.5.196>;tag=as52b812f2
Call-ID: 190e0000-a970-0f8a-8015-0002a4034048@192.168.5.152
CSeq: 10 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:92614659@192.168.5.196>
Content-Length: 0



Received SIP PDU from ( 192.168.5.196:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK19000f15a410;received=192.168.
5.152
From: <sip:406@192.168.5.196>;tag=19000f15a4
To: <sip:92614659@192.168.5.196>;tag=as52b812f2
Call-ID: 190e0000-a970-0f8a-8015-0002a4034048@192.168.5.152
CSeq: 10 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:92614659@192.168.5.196>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 1915011281 1915011282 IN IP4 192.168.5.196
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.5.196
t=0 0
m=audio 10332 RTP/AVP 18 101
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


Sending SIP PDU to ( 192.168.5.196:5060 ) from 5060
ACK sip:92614659@192.168.5.196 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK19000f15a410
From: <sip:406@192.168.5.196>;tag=19000f15a4
To: <sip:92614659@192.168.5.196>;tag=as52b812f2
Call-ID: 190e0000-a970-0f8a-8015-0002a4034048@192.168.5.152
CSeq: 10 ACK
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 192.168.5.196:5060 )
BYE sip:406@192.168.5.152 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.196:5060;branch=z9hG4bK43e18ffb;rport
Max-Forwards: 70
From: <sip:92614659@192.168.5.196>;tag=as52b812f2
To: <sip:406@192.168.5.196>;tag=19000f15a4
Call-ID: 190e0000-a970-0f8a-8015-0002a4034048@192.168.5.152
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.6
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0



Sending SIP PDU to ( 192.168.5.196:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.196:5060;branch=z9hG4bK43e18ffb;rport
From: <sip:92614659@192.168.5.196>;tag=as52b812f2
To: <sip:406@192.168.5.196>;tag=19000f15a4
Call-ID: 190e0000-a970-0f8a-8015-0002a4034048@192.168.5.152
CSeq: 102 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0



Sending SIP PDU to ( 192.168.5.196:5060 ) from 5060
REGISTER sip:192.168.5.196 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4148
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 148 REGISTER
Date: Thu, 01 Jan 1970 01:00:49 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="406", realm="asterisk", nonce="46d2bfdb", uri="s
ip:192.168.5.196", response="d55004cd3fb47b9852035e1e9c902def", algorithm=MD5
Contact: <sip:406@192.168.5.152>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 192.168.5.196:5060 )
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4148;received=192.168
.5.152
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196;tag=as3d3638b3
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 148 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e116a6c"
Content-Length: 0



Sending SIP PDU to ( 192.168.5.196:5060 ) from 5060
REGISTER sip:192.168.5.196 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4149
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 149 REGISTER
Date: Thu, 01 Jan 1970 01:00:49 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="406", realm="asterisk", nonce="5e116a6c", uri="s
ip:192.168.5.196", response="a8f64361f8411e90ca232f7acefef427", algorithm=MD5
Contact: <sip:406@192.168.5.152>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 192.168.5.196:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4149;received=192.168
.5.152
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196;tag=as3d3638b3
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 149 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:406@192.168.5.152>;expires=60
Date: Tue, 13 Apr 2010 14:29:00 GMT
Content-Length: 0


exit
Ilmira-407>
Sending SIP PDU to ( 192.168.5.196:5060 ) from 5060
REGISTER sip:192.168.5.196 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4150
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 150 REGISTER
Date: Thu, 01 Jan 1970 01:01:37 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="406", realm="asterisk", nonce="5e116a6c", uri="s
ip:192.168.5.196", response="a8f64361f8411e90ca232f7acefef427", algorithm=MD5
Contact: <sip:406@192.168.5.152>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 192.168.5.196:5060 )
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4150;received=192.168
.5.152
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196;tag=as3d065f67
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 150 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74302101"
Content-Length: 0



Sending SIP PDU to ( 192.168.5.196:5060 ) from 5060
REGISTER sip:192.168.5.196 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4151
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 151 REGISTER
Date: Thu, 01 Jan 1970 01:01:37 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="406", realm="asterisk", nonce="74302101", uri="s
ip:192.168.5.196", response="682dd7d0a0a4306a0d9eb3665b2701ec", algorithm=MD5
Contact: <sip:406@192.168.5.152>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 192.168.5.196:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4151;received=192.168
.5.152
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196;tag=as3d065f67
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 151 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:406@192.168.5.152>;expires=60
Date: Tue, 13 Apr 2010 14:29:47 GMT
Content-Length: 0



Sending SIP PDU to ( 192.168.5.196:5060 ) from 5060
REGISTER sip:192.168.5.196 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4152
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 152 REGISTER
Date: Thu, 01 Jan 1970 01:02:25 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="406", realm="asterisk", nonce="74302101", uri="s
ip:192.168.5.196", response="682dd7d0a0a4306a0d9eb3665b2701ec", algorithm=MD5
Contact: <sip:406@192.168.5.152>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 192.168.5.196:5060 )
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4152;received=192.168
.5.152
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196;tag=as18334c5f
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 152 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7479c20f"
Content-Length: 0



Sending SIP PDU to ( 192.168.5.196:5060 ) from 5060
REGISTER sip:192.168.5.196 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4153
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 153 REGISTER
Date: Thu, 01 Jan 1970 01:02:25 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="406", realm="asterisk", nonce="7479c20f", uri="s
ip:192.168.5.196", response="b1e9b8b8aa2c93ffc20f03516f7d820e", algorithm=MD5
Contact: <sip:406@192.168.5.152>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 192.168.5.196:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.152:5060;branch=z9hG4bK0a00f100a4153;received=192.168
.5.152
From: <sip:406@192.168.5.196>;tag=0a00f100a4
To: sip:406@192.168.5.196;tag=as18334c5f
Call-ID: 0a000000-e91e-f167-8000-0002a4034048@192.168.5.152
CSeq: 153 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:406@192.168.5.152>;expires=60
Date: Tue, 13 Apr 2010 14:30:35 GMT
Content-Length: 0

а это с астериска:

ast-den*CLI>
<--- SIP read from UDP:91.144.140.130:5060 --->
INVITE sip:2112166@91.144.140.130 SIP/2.0
Via: SIP/2.0/UDP 91.144.140.130;branch=91.144.140.137.5075-32770000
From: "AlterPSS" <sip:2614659@91.144.140.130>;tag=91.144.140.137.5075-282291192-14744
To: "Denis" <sip:2112166@91.144.140.130>;tag=as71134ac0
User-Agent: AlterProxySoftSwitch
Call-ID: 5a3cb3b503ffa03c3adac83b52d7e9df@91.144.140.130
CSeq: 30514 INVITE
Contact: <sip:2614659@91.144.140.130>
Content-Type: application/sdp
Content-Length: 386
Date: Tue, 13 Apr 2010 10:41:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, BYE

v=0
o=91.144.140.137 1271155260 1271155260 IN IP4 91.144.140.137
s=AlterProxySoftSwitch
c=IN IP4 91.144.140.137
t=0 0
m=image 16746 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
--- (12 headers 15 lines) ---
Sending to 91.144.140.130 : 5060 (NAT)

<--- Transmitting (NAT) to 91.144.140.130:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.144.140.130;branch=91.144.140.137.5075-32770000;received=91.144.140.130
From: "AlterPSS" <sip:2614659@91.144.140.130>;tag=91.144.140.137.5075-282291192-14744
To: "Denis" <sip:2112166@91.144.140.130>;tag=as71134ac0
Call-ID: 5a3cb3b503ffa03c3adac83b52d7e9df@91.144.140.130
CSeq: 30514 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:2112166@192.168.5.196>
Content-Length: 0


<------------>
Audio is at 192.168.5.196 port 14154
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 91.144.140.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.144.140.130;branch=91.144.140.137.5075-32770000;received=91.144.140.130
From: "AlterPSS" <sip:2614659@91.144.140.130>;tag=91.144.140.137.5075-282291192-14744
To: "Denis" <sip:2112166@91.144.140.130>;tag=as71134ac0
Call-ID: 5a3cb3b503ffa03c3adac83b52d7e9df@91.144.140.130
CSeq: 30514 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:2112166@192.168.5.196>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 774278575 774278576 IN IP4 192.168.5.196
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.5.196
t=0 0
m=audio 14154 RTP/AVP 18 101
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

<------------>
ast-den*CLI>
<--- SIP read from UDP:91.144.140.130:5060 --->
ACK sip:2112166@192.168.5.196 SIP/2.0
Via: SIP/2.0/UDP 91.144.140.130;branch=91.144.140.137.5075-32770000
From: "AlterPSS" <sip:2614659@91.144.140.130>;tag=91.144.140.137.5075-282291192-14744
To: "Denis" <sip:2112166@91.144.140.130>;tag=as71134ac0
Call-ID: 5a3cb3b503ffa03c3adac83b52d7e9df@91.144.140.130
CSeq: 30514 ACK
Contact: <sip:2614659@91.144.140.130>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '0a000000-e91e-f167-8000-0002a4034048@192.168.5.152' Method: REGISTER
ast-den*CLI>
<--- SIP read from UDP:91.144.140.130:5060 --->
BYE sip:2112166@192.168.5.196 SIP/2.0
Via: SIP/2.0/UDP 91.144.140.130;branch=91.144.140.137.5075-33770000
From: "AlterPSS" <sip:2614659@91.144.140.130>;tag=91.144.140.137.5075-282291192-14744
To: "Denis" <sip:2112166@91.144.140.130>;tag=as71134ac0
Call-ID: 5a3cb3b503ffa03c3adac83b52d7e9df@91.144.140.130
CSeq: 30515 BYE
Content-Length: 0
Date: Tue, 13 Apr 2010 10:41:32 GMT


<------------->
--- (8 headers 0 lines) ---
Sending to 91.144.140.130 : 5060 (NAT)

<--- Transmitting (NAT) to 91.144.140.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.144.140.130;branch=91.144.140.137.5075-33770000;received=91.144.140.130
From: "AlterPSS" <sip:2614659@91.144.140.130>;tag=91.144.140.137.5075-282291192-14744
To: "Denis" <sip:2112166@91.144.140.130>;tag=as71134ac0
Call-ID: 5a3cb3b503ffa03c3adac83b52d7e9df@91.144.140.130
CSeq: 30515 BYE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
== Spawn extension (incoming, 92614659, 1) exited non-zero on 'SIP/406-00000008'
Scheduling destruction of SIP dialog '53100000-38c6-2694-801b-0002a4034048@192.168.5.152' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:406@192.168.5.152> for address/port to send to
set_destination: set destination to 192.168.5.152, port 5060
Reliably Transmitting (no NAT) to 192.168.5.152:5060:
BYE sip:406@192.168.5.152 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.196:5060;branch=z9hG4bK27b85029;rport
Max-Forwards: 70
From: <sip:92614659@192.168.5.196>;tag=as7349df29
To: <sip:406@192.168.5.196>;tag=5300261ba4
Call-ID: 53100000-38c6-2694-801b-0002a4034048@192.168.5.152
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.6
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
ast-den*CLI>
<--- SIP read from UDP:192.168.5.152:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.196:5060;branch=z9hG4bK27b85029;rport
From: <sip:92614659@192.168.5.196>;tag=as7349df29
To: <sip:406@192.168.5.196>;tag=5300261ba4
Call-ID: 53100000-38c6-2694-801b-0002a4034048@192.168.5.152
CSeq: 102 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '5a3cb3b503ffa03c3adac83b52d7e9df@91.144.140.130' Method: BYE
Really destroying SIP dialog '53100000-38c6-2694-801b-0002a4034048@192.168.5.152' Method: ACK

Идеал - это недостижимая мечта.. :)

Логи тоже надо уметь читать, а опыта в астериксе такого нет к сожалению