Здрасть! Хочу поднять уже замусоленную тему - астериск и факсы. Есть оконечное устройство - шлюз Mediatrix 1104-SIP, к нему прицеплен факс. Есть второй факс - в ТфОП. Так же есть киска с включенным в нее потоком Е1. Поток этот приходит от АТС. И есть собсна сам астериск.
Пытаюсь отправить - между факсами не устанавливается соединение.

Конфиги киски:
voice service voip
fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback cisco
!
voice class codec 777
codec preference 1 g711alaw bytes 240
codec preference 2 g711ulaw bytes 240
!
controller E1 3/0
framing NO-CRC4
clock source line primary
pri-group timeslots 1-31
!
interface Serial3/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn T309-enable
isdn negotiate-bchan
isdn incoming progress validate
no fair-queue
no cdp enable
!
voice-port 3/0:15
!
dial-peer voice 1 voip
incoming called-number .T
destination-pattern .T
modem passthrough nse codec g711alaw
voice-class codec 777
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay h245-signal
playout-delay nominal 40
playout-delay mode fixed
fax rate 14400
fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback cisco
ip qos dscp cs5 media
ip qos dscp cs5 signaling
no vad
!
dial-peer voice 2 pots
destination-pattern 777#.T
translate-outgoing called 777
direct-inward-dial
port 3/0:15
!
sip-ua
sip-server ipv4:[asterisk_ip]
!

Конфиг астериска:
[general]
context=default
allowguest=yes
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
autocreatepeer=yes
sipdebug=no
musicclass=default
; language=ru
canreinvite=no
disallow=all
t38pt_udptl=yes
allow=alaw,ulaw
tos_sip=cs3
tos_audio=ef

[200] ; Planet VIP-153T
type=friend
host=dynamic
username=200
secret=xxx
nat=yes
context=kvant-4users
callerid="Planet VIP-153T" <96190>

[cisco]
type=peer
host=[cisco_ip]
context=default

Логи попытки передачи факса дальше дальше.

Астериск [часть 1]:
<--- SIP read from [phone_ip]:10000 --->
INVITE sip:97112@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK28f73e243
Content-Length: 239
To: sip:97112@asterisk
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617984 INVITE
Route: <sip:[asterisk_ip]:5060;lr>
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: 96191 <sip:201@192.168.0.13:5060>
Supported: replaces
User-Agent: MxSipApp/5.0.15.90 MxSF/v3.2.1.1

v=0
o=MxSIP 919670949189526481 722299189156192730 IN IP4 192.168.0.13
s=-
c=IN IP4 192.168.0.13
t=0 0
a=sendrecv
m=audio 5004 RTP/AVP 8 18 4 0
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000

<------------->
--- (16 headers 11 lines) ---
**** Received INVITE (5) - Command in SIP INVITE
Sending to 192.168.0.13 : 5060 (no NAT)
Using INVITE request as basis request - 2288a8b02522ac2b9e312ee06118e4d2@asterisk
Found user '201' for '201'
SoftSwitch*CLI>
<--- Reliably Transmitting (NAT) to [phone_ip]:10000 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK28f73e243;received=[phone_ip]
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
To: sip:97112@asterisk;tag=as22725678
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617984 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22ebaa37"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2288a8b02522ac2b9e312ee06118e4d2@asterisk' in 32000 ms (Method: INVITE)
SoftSwitch*CLI>
<--- SIP read from [phone_ip]:10000 --->
ACK sip:97112@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK28f73e243
Content-Length: 0
To: sip:97112@asterisk;tag=as22725678
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617984 ACK
Route: <sip:[asterisk_ip]:5060;lr>
User-Agent: MxSipApp/5.0.15.90 MxSF/v3.2.1.1


<------------->
--- (9 headers 0 lines) ---
**** Received ACK (6) - Command in SIP ACK
setting state to INV_CONFIRMED
SoftSwitch*CLI>
<--- SIP read from [phone_ip]:10000 --->
INVITE sip:97112@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK76da58e8c
Content-Length: 239
To: sip:97112@asterisk
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617985 INVITE
Route: <sip:[asterisk_ip]:5060;lr>
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Contact: 96191 <sip:201@192.168.0.13:5060>
Content-Type: application/sdp
Authorization:Digest response="de72125c10181c105db7e04c0513ddae",username="201",realm="asterisk",nonce="22ebaa37",algorithm=MD5,uri="sip:97112@asterisk"
Supported: replaces
User-Agent: MxSipApp/5.0.15.90 MxSF/v3.2.1.1

v=0
o=MxSIP 919670949189526481 722299189156192730 IN IP4 192.168.0.13
s=-
c=IN IP4 192.168.0.13
t=0 0
a=sendrecv
m=audio 5004 RTP/AVP 8 18 4 0
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000

<------------->
--- (17 headers 11 lines) ---
**** Received INVITE (5) - Command in SIP INVITE
Sending to [phone_ip] : 10000 (NAT)
Using INVITE request as basis request - 2288a8b02522ac2b9e312ee06118e4d2@asterisk
Found user '201' for '201'
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Peer audio RTP is at port 192.168.0.13:5004
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format G723 for ID 4
Found description format PCMU for ID 0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.13:5004
Looking for 97112 in kvant-4users (domain asterisk)
list_route: hop: <sip:201@192.168.0.13:5060>
SoftSwitch*CLI>
<--- Transmitting (NAT) to [phone_ip]:10000 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK76da58e8c;received=[phone_ip]
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
To: sip:97112@asterisk
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617985 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97112@[asterisk_ip]>
Content-Length: 0


<------------>
Audio is at [asterisk_ip] port 13486
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to [cisco_ip]:5060:
INVITE sip:777#84843997112@[cisco_ip] SIP/2.0
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK1a10c5cb;rport
Max-Forwards: 70
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as4686ba72
To: <sip:777#84843997112@[cisco_ip]>
Contact: <sip:96191@[asterisk_ip]>
Call-ID: 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 20 Dec 2006 15:34:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 10196 10196 IN IP4 [asterisk_ip]
s=session
c=IN IP4 [asterisk_ip]
t=0 0
m=audio 13486 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
SoftSwitch*CLI>
<--- Transmitting (NAT) to [phone_ip]:10000 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK76da58e8c;received=[phone_ip]
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
To: sip:97112@asterisk;tag=as5f0cbffe
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617985 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97112@[asterisk_ip]>
Content-Length: 0


<------------>
SoftSwitch*CLI>
<--- SIP read from [cisco_ip]:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK1a10c5cb;rport
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as4686ba72
To: <sip:777#84843997112@[cisco_ip]>;tag=127445C-207C
Date: Mon, 01 Mar 1993 05:22:30 GMT
Call-ID: 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
SoftSwitch*CLI>
<--- SIP read from [cisco_ip]:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK1a10c5cb;rport
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as4686ba72
To: <sip:777#84843997112@[cisco_ip]>;tag=127445C-207C
Date: Mon, 01 Mar 1993 05:22:30 GMT
Call-ID: 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:777#84843997112@[cisco_ip]:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 197

v=0
o=CiscoSystemsSIP-GW-UserAgent 4399 4101 IN IP4 [cisco_ip]
s=SIP Call
c=IN IP4 [cisco_ip]
t=0 0
m=audio 19080 RTP/AVP 8
c=IN IP4 [cisco_ip]
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (13 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port [cisco_ip]:19080
Found description format PCMA for ID 8
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port [cisco_ip]:19080
SoftSwitch*CLI>
<--- SIP read from [cisco_ip]:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK1a10c5cb;rport
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as4686ba72
To: <sip:777#84843997112@[cisco_ip]>;tag=127445C-207C
Date: Mon, 01 Mar 1993 05:22:30 GMT
Call-ID: 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:777#84843997112@[cisco_ip]:5060>
Content-Type: application/sdp
Content-Length: 197

v=0
o=CiscoSystemsSIP-GW-UserAgent 4399 4101 IN IP4 [cisco_ip]
s=SIP Call
c=IN IP4 [cisco_ip]
t=0 0
m=audio 19080 RTP/AVP 8
c=IN IP4 [cisco_ip]
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (13 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port [cisco_ip]:19080
Found description format PCMA for ID 8
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port [cisco_ip]:19080
--- set_address_from_contact host '[cisco_ip]'
list_route: hop: <sip:777#84843997112@[cisco_ip]:5060>
transmit_request ACK
set_destination: Parsing <sip:777#84843997112@[cisco_ip]:5060> for address/port to send to
set_destination: set destination to [cisco_ip], port 5060
Transmitting (no NAT) to [cisco_ip]:5060:
ACK sip:777#84843997112@[cisco_ip]:5060 SIP/2.0
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK3e232a0c;rport
Max-Forwards: 70
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as4686ba72
To: <sip:777#84843997112@[cisco_ip]>;tag=127445C-207C
Contact: <sip:96191@[asterisk_ip]>
Call-ID: 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
Audio is at [asterisk_ip] port 19606
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP

Астериск [часть 2]:
<--- SIP read from [phone_ip]:10000 --->
ACK sip:97112@[asterisk_ip] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK56daeb2b9
Content-Length: 0
To: sip:97112@asterisk;tag=as5f0cbffe
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617985 ACK
Contact: 96191 <sip:201@192.168.0.13:5060>
Authorization:Digest response="de72125c10181c105db7e04c0513ddae",username="201",realm="asterisk",nonce="22ebaa37",algorithm=MD5,uri="sip:97112@asterisk"
User-Agent: MxSipApp/5.0.15.90 MxSF/v3.2.1.1
<------------->
**** Received ACK (6) - Command in SIP ACK
setting state to INV_CONFIRMED
<--- SIP read from [phone_ip]:10000 --->
INVITE sip:97112@[asterisk_ip] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK25f2d6d3f
Content-Length: 335
To: sip:97112@asterisk;tag=as5f0cbffe
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617986 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: 96191 <sip:201@192.168.0.13:5060>
Authorization:Digest response="50c779a163daf6bd48d4d53f350b2b24",username="201",realm="asterisk",nonce="22ebaa37",algorithm=MD5,uri="sip:97112@[asterisk_ip]"
Supported: replaces
User-Agent: MxSipApp/5.0.15.90 MxSF/v3.2.1.1

v=0
o=MxSIP 919670949189526481 722299189156192731 IN IP4 192.168.0.13
s=-
c=IN IP4 192.168.0.13
t=0 0
a=sendrecv
m=image 6004 udptl t38
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxVersion:0
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
**** Received INVITE (5) - Command in SIP INVITE
Sending to [phone_ip] : 10000 (NAT)
Got T.38 offer in SDP in dialog 2288a8b02522ac2b9e312ee06118e4d2@asterisk
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 2288a8b02522ac2b9e312ee06118e4d2@asterisk
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:777#84843997112@[cisco_ip]:5060> for address/port to send to
set_destination: set destination to [cisco_ip], port 5060
Reliably Transmitting (no NAT) to [cisco_ip]:5060:
INVITE sip:777#84843997112@[cisco_ip]:5060 SIP/2.0
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK2dbcd3e0;rport
Max-Forwards: 70
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as4686ba72
To: <sip:777#84843997112@[cisco_ip]>;tag=127445C-207C
Contact: <sip:96191@[asterisk_ip]>
Call-ID: 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 10196 10197 IN IP4 [asterisk_ip]
s=session
c=IN IP4 [asterisk_ip]
t=0 0
m=image 4567 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

<--- SIP read from [cisco_ip]:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK2dbcd3e0;rport
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as4686ba72
To: <sip:777#84843997112@[cisco_ip]>;tag=127445C-207C
Date: Mon, 01 Mar 1993 05:23:15 GMT
Call-ID: 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:777#84843997112@[cisco_ip]:5060>
Content-Type: application/sdp
Content-Length: 163

v=0
o=CiscoSystemsSIP-GW-UserAgent 4399 4102 IN IP4 [cisco_ip]
s=SIP Call
c=IN IP4 [cisco_ip]
t=0 0
m=image 19080 udptl t38
c=IN IP4 [cisco_ip]

<------------->
Got T.38 offer in SDP in dialog 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
--- set_address_from_contact host '[cisco_ip]'

<--- Reliably Transmitting (NAT) to [phone_ip]:10000 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK25f2d6d3f;received=[phone_ip]
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
To: sip:97112@asterisk;tag=as5f0cbffe
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617986 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97112@[asterisk_ip]>
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 10196 10197 IN IP4 [asterisk_ip]
s=session
c=IN IP4 [asterisk_ip]
t=0 0
m=image 4333 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

<------------>
transmit_request ACK
set_destination: Parsing <sip:777#84843997112@[cisco_ip]:5060> for address/port to send to
set_destination: set destination to [cisco_ip], port 5060
Transmitting (no NAT) to [cisco_ip]:5060:
ACK sip:777#84843997112@[cisco_ip]:5060 SIP/2.0
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK28ff4951;rport
Max-Forwards: 70
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as4686ba72
To: <sip:777#84843997112@[cisco_ip]>;tag=127445C-207C
Contact: <sip:96191@[asterisk_ip]>
Call-ID: 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<--- SIP read from [phone_ip]:10000 --->
ACK sip:97112@[asterisk_ip] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bKcbd9c6802
Content-Length: 0
To: sip:97112@asterisk;tag=as5f0cbffe
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617986 ACK
Contact: 96191 <sip:201@192.168.0.13:5060>
Authorization:Digest response="50c779a163daf6bd48d4d53f350b2b24",username="201",realm="asterisk",nonce="22ebaa37",algorithm=MD5,uri="sip:97112@[asterisk_ip]"
User-Agent: MxSipApp/5.0.15.90 MxSF/v3.2.1.1

<------------->
**** Received ACK (6) - Command in SIP ACK
setting state to INV_CONFIRMED
SoftSwitch*CLI>
<--- SIP read from [phone_ip]:10000 --->
BYE sip:97112@[asterisk_ip] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK0309ca4e4
Content-Length: 0
To: sip:97112@asterisk;tag=as5f0cbffe
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617987 BYE
Supported: timer
Authorization:Digest response="0930d8b7b4e141547b33b42183f2665e",username="201",realm="asterisk",nonce="22ebaa37",algorithm=MD5,uri="sip:97112@[asterisk_ip]"
Supported: replaces
User-Agent: MxSipApp/5.0.15.90 MxSF/v3.2.1.1

<------------->
**** Received BYE (8) - Command in SIP BYE
Sending to [phone_ip] : 10000 (NAT)
<--- Transmitting (NAT) to [phone_ip]:10000 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK0309ca4e4;received=[phone_ip]
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
To: sip:97112@asterisk;tag=as5f0cbffe
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617987 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97112@[asterisk_ip]>
Content-Length: 0


<------------>
chan_sip1 sip_hangup flags invitestate 5 0x8042004 data <INVITE>
set_destination: Parsing <sip:777#84843997112@[cisco_ip]:5060> for address/port to send to
set_destination: set destination to [cisco_ip], port 5060
Reliably Transmitting (no NAT) to [cisco_ip]:5060:
BYE sip:777#84843997112@[cisco_ip]:5060 SIP/2.0
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK20c8afda;rport
Max-Forwards: 70
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as4686ba72
To: <sip:777#84843997112@[cisco_ip]>;tag=127445C-207C
Call-ID: 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Content-Length: 0


---
chan_sip1 sip_hangup flags now 0x8042004
chan_sip1 sip_hangup flags invitestate 6 0x80c000d data <BYE>
chan_sip1 sip_hangup flags now 0x80c000f
<--- SIP read from [cisco_ip]:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK20c8afda;rport
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as4686ba72
To: <sip:777#84843997112@[cisco_ip]>;tag=127445C-207C
Date: Mon, 01 Mar 1993 05:23:15 GMT
Call-ID: 4ba15939234fc4713c8bdd55023f3983@[asterisk_ip]
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 104 BYE

<--- Reliably Transmitting (NAT) to [phone_ip]:10000 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK76da58e8c;received=[phone_ip]
From: 96191 <sip:201@asterisk>;tag=ec0aceb029687d1
To: sip:97112@asterisk;tag=as5f0cbffe
Call-ID: 2288a8b02522ac2b9e312ee06118e4d2@asterisk
CSeq: 1977617985 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97112@[asterisk_ip]>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 10196 10196 IN IP4 [asterisk_ip]
s=session
c=IN IP4 [asterisk_ip]
t=0 0
m=audio 19606 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

друже, воспользуйся pastebin.ca и сюда ссылку кидай.
Так будет имхо правильнее и красивее.

киска (ccsip messages + isdn):
*Mar 1 05:19:16.946: Received:
INVITE sip:777#84843976702@[cisco_ip] SIP/2.0
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK1199fdda;rport
Max-Forwards: 70
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as7340675a
To: <sip:777#84843976702@[cisco_ip]>
Contact: <sip:96191@[asterisk_ip]>
Call-ID: 4fd74c2117a2ae531b9d46f92de0ca3b@[asterisk_ip]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 20 Dec 2006 15:31:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 10196 10196 IN IP4 [asterisk_ip]
s=session
c=IN IP4 [asterisk_ip]
t=0 0
m=audio 19130 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

*Mar 1 05:19:16.954: ISDN Se3/0:15 Q931: TX -> SETUP pd = 8 callref = 0x00A7
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA1839F
Preferred, Channel 31
Calling Party Number i = 0x0080, '96191'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '84843976702'
Plan:Unknown, Type:Unknown
*Mar 1 05:19:16.954: Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK1199fdda;rport
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as7340675a
To: <sip:777#84843976702@[cisco_ip]>;tag=1244FD8-10FC
Date: Mon, 01 Mar 1993 05:19:16 GMT
Call-ID: 4fd74c2117a2ae531b9d46f92de0ca3b@[asterisk_ip]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0



*Mar 1 05:19:17.050: ISDN Se3/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x80A7
Channel ID i = 0xA9839F
Exclusive, Channel 31
*Mar 1 05:19:19.402: ISDN Se3/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x80A7
*Mar 1 05:19:19.438: ISDN Se3/0:15 Q931: RX <- PROGRESS pd = 8 callref = 0x80A7
Progress Ind i = 0x8488 - In-band info or appropriate now available
Progress Ind i = 0x8281 - Call not end-to-end ISDN, may have in-band info
*Mar 1 05:19:19.442: Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK1199fdda;rport
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as7340675a
To: <sip:777#84843976702@[cisco_ip]>;tag=1244FD8-10FC
Date: Mon, 01 Mar 1993 05:19:16 GMT
Call-ID: 4fd74c2117a2ae531b9d46f92de0ca3b@[asterisk_ip]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:777#84843976702@[cisco_ip]:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 195

v=0
o=CiscoSystemsSIP-GW-UserAgent 163 535 IN IP4 [cisco_ip]
s=SIP Call
c=IN IP4 [cisco_ip]
t=0 0
m=audio 16476 RTP/AVP 8
c=IN IP4 [cisco_ip]
a=rtpmap:8 PCMA/8000
a=ptime:20

*Mar 1 05:19:27.298: ISDN Se3/0:15 Q931: RX <- CONNECT pd = 8 callref = 0x80A7
Progress Ind i = 0x8281 - Call not end-to-end ISDN, may have in-band info
*Mar 1 05:19:27.298: %ISDN-6-CONNECT: Interface Serial3/0:30 is now connected to 84843976702 N/A
*Mar 1 05:19:27.298: ISDN Se3/0:15 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x00A7
*Mar 1 05:19:27.302: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK1199fdda;rport
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as7340675a
To: <sip:777#84843976702@[cisco_ip]>;tag=1244FD8-10FC
Date: Mon, 01 Mar 1993 05:19:16 GMT
Call-ID: 4fd74c2117a2ae531b9d46f92de0ca3b@[asterisk_ip]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:777#84843976702@[cisco_ip]:5060>
Content-Type: application/sdp
Content-Length: 195

v=0
o=CiscoSystemsSIP-GW-UserAgent 163 535 IN IP4 [cisco_ip]
s=SIP Call
c=IN IP4 [cisco_ip]
t=0 0
m=audio 16476 RTP/AVP 8
c=IN IP4 [cisco_ip]
a=rtpmap:8 PCMA/8000
a=ptime:20

*Mar 1 05:19:27.302: Received:
ACK sip:777#84843976702@[cisco_ip]:5060 SIP/2.0
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK61ea61ce;rport
Max-Forwards: 70
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as7340675a
To: <sip:777#84843976702@[cisco_ip]>;tag=1244FD8-10FC
Contact: <sip:96191@[asterisk_ip]>
Call-ID: 4fd74c2117a2ae531b9d46f92de0ca3b@[asterisk_ip]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0



*Mar 1 05:21:36.762: Received:
INVITE sip:777#84843976702@[cisco_ip]:5060 SIP/2.0
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK631e2a14;rport
Max-Forwards: 70
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as7340675a
To: <sip:777#84843976702@[cisco_ip]>;tag=1244FD8-10FC
Contact: <sip:96191@[asterisk_ip]>
Call-ID: 4fd74c2117a2ae531b9d46f92de0ca3b@[asterisk_ip]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 10196 10197 IN IP4 [asterisk_ip]
s=session
c=IN IP4 [asterisk_ip]
t=0 0
m=image 4972 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

*Mar 1 05:21:36.762: Received:
erredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy


*Mar 1 05:21:36.766: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK631e2a14;rport
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as7340675a
To: <sip:777#84843976702@[cisco_ip]>;tag=1244FD8-10FC
Date: Mon, 01 Mar 1993 05:21:36 GMT
Call-ID: 4fd74c2117a2ae531b9d46f92de0ca3b@[asterisk_ip]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:777#84843976702@[cisco_ip]:5060>
Content-Type: application/sdp
Content-Length: 161

v=0
o=CiscoSystemsSIP-GW-UserAgent 163 536 IN IP4 [cisco_ip]
s=SIP Call
c=IN IP4 [cisco_ip]
t=0 0
m=image 16476 udptl t38
c=IN IP4 [cisco_ip]

*Mar 1 05:21:36.770: Received:
ACK sip:777#84843976702@[cisco_ip]:5060 SIP/2.0
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK5f927316;rport
Max-Forwards: 70
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as7340675a
To: <sip:777#84843976702@[cisco_ip]>;tag=1244FD8-10FC
Contact: <sip:96191@[asterisk_ip]>
Call-ID: 4fd74c2117a2ae531b9d46f92de0ca3b@[asterisk_ip]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0



*Mar 1 05:21:37.002: Received:
BYE sip:777#84843976702@[cisco_ip]:5060 SIP/2.0
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK17fcdbbe;rport
Max-Forwards: 70
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as7340675a
To: <sip:777#84843976702@[cisco_ip]>;tag=1244FD8-10FC
Call-ID: 4fd74c2117a2ae531b9d46f92de0ca3b@[asterisk_ip]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Content-Length: 0



*Mar 1 05:21:37.006: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP [asterisk_ip]:5060;branch=z9hG4bK17fcdbbe;rport
From: "gtw-1" <sip:96191@[asterisk_ip]>;tag=as7340675a
To: <sip:777#84843976702@[cisco_ip]>;tag=1244FD8-10FC
Date: Mon, 01 Mar 1993 05:21:37 GMT
Call-ID: 4fd74c2117a2ae531b9d46f92de0ca3b@[asterisk_ip]
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 104 BYE



*Mar 1 05:21:37.514: %ISDN-6-DISCONNECT: Interface Serial3/0:30 disconnected from 84843976702 , call lasted 130 seconds
*Mar 1 05:21:37.514: ISDN Se3/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x00A7
Cause i = 0x8090 - Normal call clearing
*Mar 1 05:21:37.566: ISDN Se3/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x80A7
*Mar 1 05:21:37.566: ISDN Se3/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x00A7

Сначала написал, потом заметил --- <sip:96191@> нигде домен не написан. В логах всё ок, это я коряво заменил.