и снова Т38...

[Psyhoed]

Всем хорошего настроения! Помогите немного начинающему экспериментатору, плиз
Имеется Linksys SPA2102, * 10.3.0 и Cisco AS5300 (IOS 12.3(26)). схема следующая:
fax1---SPA2102 ----T38,sip------ eth1(asterisk)eth0-----t38,sip---- as5300 ---E1--- Avaya Definity --- fax2 т.е. астериск слушает 2 сетки.
Голосовые вызовы ходят везде, тут все хорошо.
На астериске создан экстеншн, на который отвечает RecеiveFAX. fax1 - asterisk все летает на ура по Т38, вот из дебага (апишники засекурил малость:) последний байт=17 эт астрериск, а 18 Linksys)

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from UDP:хх.хх.хх.18:5060 --->
INVITE sip:3016@хх.хх.хх.17 SIP/2.0
Via: SIP/2.0/UDP хх.хх.хх.18:5060;branch=z9hG4bK-cd2e84b
From: 1350 <sip:1350@хх.хх.хх.17>;tag=c26e94a38e6c2507o1
To: <sip:3016@хх.хх.хх.17>
Remote-Party-ID: 1350 <sip:1350@хх.хх.хх.17>;screen=yes;party=calling
Call-ID: cf96d1d3-36c58997@хх.хх.хх.18
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="1350",realm="asterisk",nonce="2aed2eef",uri="sip:3016@хх.хх.хх.17",algorithm=MD5,response="b7cacb7ff548e3161bfabc835421164a"
Contact: 1350 <sip:1350@хх.хх.хх.18:5060>
Expires: 240
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 259
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 6469327 6469327 IN IP4 хх.хх.хх.18
s=-
c=IN IP4 хх.хх.хх.18
t=0 0
m=audio 16418 RTP/AVP 18 100 101
a=rtpmap:18 G729/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (16 headers 13 lines) ---
Sending to хх.хх.хх.18:5060 (no NAT)
Using INVITE request as basis request - cf96d1d3-36c58997@хх.хх.хх.18
Found peer '1350' for '1350' from хх.хх.хх.18:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Found audio description format G729 for ID 18
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port хх.хх.хх.18:16418
Looking for 3016 in default (domain хх.хх.хх.17)
list_route: hop: <sip:1350@хх.хх.хх.18:5060>

<--- Transmitting (no NAT) to хх.хх.хх.18:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP хх.хх.хх.18:5060;branch=z9hG4bK-cd2e84b;received=хх.хх.хх.18
From: 1350 <sip:1350@хх.хх.хх.17>;tag=c26e94a38e6c2507o1
To: <sip:3016@хх.хх.хх.17>
Call-ID: cf96d1d3-36c58997@хх.хх.хх.18
CSeq: 102 INVITE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3016@хх.хх.хх.17:5060>
Content-Length: 0


<------------>
-- Executing [3016@default:1] Answer("SIP/1350-0000004a", "") in new stack
Audio is at 13150
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to хх.хх.хх.18:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP хх.хх.хх.18:5060;branch=z9hG4bK-cd2e84b;received=хх.хх.хх.18
From: 1350 <sip:1350@хх.хх.хх.17>;tag=c26e94a38e6c2507o1
To: <sip:3016@хх.хх.хх.17>;tag=as36d0f6b1
Call-ID: cf96d1d3-36c58997@хх.хх.хх.18
CSeq: 102 INVITE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3016@хх.хх.хх.17:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1348915240 1348915240 IN IP4 хх.хх.хх.17
s=Asterisk PBX 10.3.0
c=IN IP4 хх.хх.хх.17
t=0 0
m=audio 13150 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=ptime:20
a=sendrecv

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

<--- SIP read from UDP:хх.хх.хх.18:5060 --->
ACK sip:3016@хх.хх.хх.17:5060 SIP/2.0
Via: SIP/2.0/UDP хх.хх.хх.18:5060;branch=z9hG4bK-3237a581
From: 1350 <sip:1350@хх.хх.хх.17>;tag=c26e94a38e6c2507o1
To: <sip:3016@хх.хх.хх.17>;tag=as36d0f6b1
Call-ID: cf96d1d3-36c58997@хх.хх.хх.18
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="1350",realm="asterisk",nonce="2aed2eef",uri="sip:3016@хх.хх.хх.17:5060",algorithm=MD5,response="141a11ba1e8ab021dd8806e3153f0101"
Contact: 1350 <sip:1350@хх.хх.хх.18:5060>
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
-- Executing [3016@default:2] Set("SIP/1350-0000004a", "DT=1335437761.74") in new stack
-- Executing [3016@default:3] ReceiveFAX("SIP/1350-0000004a", "/var/spool/asterisk/fax/1335437761.74.tif") in new stack
-- Channel 'SIP/1350-0000004a' receiving FAX '/var/spool/asterisk/fax/1335437761.74.tif'
== Using UDPTL CoS mark 5
set_destination: Parsing <sip:1350@хх.хх.хх.18:5060> for address/port to send to
set_destination: set destination to хх.хх.хх.18:5060
Reliably Transmitting (no NAT) to хх.хх.хх.18:5060:
INVITE sip:1350@хх.хх.хх.18:5060 SIP/2.0
Via: SIP/2.0/UDP хх.хх.хх.17:5060;branch=z9hG4bK0918a80e
Max-Forwards: 70
From: <sip:3016@хх.хх.хх.17>;tag=as36d0f6b1
To: 1350 <sip:1350@хх.хх.хх.17>;tag=c26e94a38e6c2507o1
Contact: <sip:3016@хх.хх.хх.17:5060>
Call-ID: cf96d1d3-36c58997@хх.хх.хх.18
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.0
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: 273

v=0
o=root 1348915240 1348915241 IN IP4 хх.хх.хх.17
s=Asterisk PBX 10.3.0
c=IN IP4 хх.хх.хх.17
t=0 0
m=image 4868 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

---

<--- SIP read from UDP:хх.хх.хх.18:5060 --->
SIP/2.0 200 OK
To: 1350 <sip:1350@хх.хх.хх.17>;tag=c26e94a38e6c2507o1
From: <sip:3016@хх.хх.хх.17>;tag=as36d0f6b1
Call-ID: cf96d1d3-36c58997@хх.хх.хх.18
CSeq: 102 INVITE
Via: SIP/2.0/UDP хх.хх.хх.17:5060;branch=z9hG4bK0918a80e
Contact: 1350 <sip:1350@хх.хх.хх.18:5060>
Server: Linksys/SPA2102-3.3.6
Content-Length: 269
Content-Type: application/sdp

v=0
o=- 6469683 6469683 IN IP4 хх.хх.хх.18
s=-
c=IN IP4 хх.хх.хх.18
t=0 0
m=image 16418 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (10 headers 12 lines) ---
Got T.38 offer in SDP in dialog cf96d1d3-36c58997@хх.хх.хх.18
Capabilities: us - (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:1350@хх.хх.хх.18:5060> for address/port to send to
set_destination: set destination to хх.хх.хх.18:5060
Transmitting (no NAT) to хх.хх.хх.18:5060:
ACK sip:1350@хх.хх.хх.18:5060 SIP/2.0
Via: SIP/2.0/UDP хх.хх.хх.17:5060;branch=z9hG4bK2e797e12
Max-Forwards: 70
From: <sip:3016@хх.хх.хх.17>;tag=as36d0f6b1
To: 1350 <sip:1350@хх.хх.хх.17>;tag=c26e94a38e6c2507o1
Contact: <sip:3016@хх.хх.хх.17:5060>
Call-ID: cf96d1d3-36c58997@хх.хх.хх.18
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.3.0
Content-Length: 0

а вот fax2 - asterisk нифига соответственно fax1 - fax2 и наоборот тож.

дебаг в сторону циски:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from UDP:192.168.10.2:56979 --->
INVITE sip:3016@192.168.10.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060
From: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
To: <sip:3016@192.168.10.1>
Date: Wed, 25 Jan 2006 19:41:36 GMT
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 1846746907-2366706138-2240580035-1638853025
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: <sip:2222@192.168.10.2>;party=calling;screen=no;privacy=off
Timestamp: 1138218096
Contact: <sip:2222@192.168.10.2:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 215

v=0
o=CiscoSystemsSIP-GW-UserAgent 8524 3810 IN IP4 192.168.10.2
s=SIP Call
c=IN IP4 192.168.10.2
t=0 0
m=audio 18912 RTP/AVP 8 19
c=IN IP4 192.168.10.2
a=rtpmap:8 PCMA/8000
a=rtpmap:19 CN/8000
a=ptime:20
<------------->
--- (20 headers 10 lines) ---
Sending to 192.168.10.2:56979 (NAT)
Using INVITE request as basis request - 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
Found peer 'as5300' for '2222' from 192.168.10.2:56979
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 19
Found audio description format PCMA for ID 8
Found audio description format CN for ID 19
Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x2 (CN|), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.10.2:18912
Looking for 3016 in from-as5300 (domain 192.168.10.1)
list_route: hop: <sip:2222@192.168.10.2:5060>

<--- Transmitting (NAT) to 192.168.10.2:56979 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.2:5060;received=192.168.10.2;rport=56979
From: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
To: <sip:3016@192.168.10.1>
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
CSeq: 101 INVITE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3016@192.168.10.1:5060>
Content-Length: 0


<------------>
-- Executing [3016@from-as5300:1] Answer("SIP/as5300-0000006c", "") in new stack
Audio is at 11956
Adding codec 100004 (alaw) to SDP

<--- Reliably Transmitting (NAT) to 192.168.10.2:56979 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.2:5060;received=192.168.10.2;rport=56979
From: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
To: <sip:3016@192.168.10.1>;tag=as3f705421
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
CSeq: 101 INVITE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3016@192.168.10.1:5060>
Content-Type: application/sdp
Content-Length: 177

v=0
o=root 612798355 612798355 IN IP4 192.168.10.1
s=Asterisk PBX 10.3.0
c=IN IP4 192.168.10.1
t=0 0
m=audio 11956 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:192.168.10.2:52492 --->
ACK sip:3016@192.168.10.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060
From: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
To: <sip:3016@192.168.10.1>;tag=as3f705421
Date: Wed, 25 Jan 2006 19:41:36 GMT
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK

<------------->
--- (9 headers 0 lines) ---
-- Executing [3016@from-as5300:2] Set("SIP/as5300-0000006c", "DT=1335445600.108") in new stack
-- Executing [3016@from-as5300:3] ReceiveFAX("SIP/as5300-0000006c", "/var/spool/asterisk/fax/1335445600.108.tif") in new stack
-- Channel 'SIP/as5300-0000006c' receiving FAX '/var/spool/asterisk/fax/1335445600.108.tif'
== Using UDPTL CoS mark 5
set_destination: Parsing <sip:2222@192.168.10.2:5060> for address/port to send to
set_destination: set destination to 192.168.10.2:5060
Reliably Transmitting (NAT) to 192.168.10.2:52492:
INVITE sip:2222@192.168.10.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0e0422b7;rport
Max-Forwards: 70
From: <sip:3016@192.168.10.1>;tag=as3f705421
To: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
Contact: <sip:3016@192.168.10.1:5060>
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.0
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: 269

v=0
o=root 612798355 612798356 IN IP4 192.168.10.1
s=Asterisk PBX 10.3.0
c=IN IP4 192.168.10.1
t=0 0
m=image 4369 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

---
Retransmitting #1 (NAT) to 192.168.10.2:52492:
INVITE sip:2222@192.168.10.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0e0422b7;rport
Max-Forwards: 70
From: <sip:3016@192.168.10.1>;tag=as3f705421
To: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
Contact: <sip:3016@192.168.10.1:5060>
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.0
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: 269

v=0
o=root 612798355 612798356 IN IP4 192.168.10.1
s=Asterisk PBX 10.3.0
c=IN IP4 192.168.10.1
t=0 0
m=image 4369 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

---
Retransmitting #2 (NAT) to 192.168.10.2:52492:
INVITE sip:2222@192.168.10.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0e0422b7;rport
Max-Forwards: 70
From: <sip:3016@192.168.10.1>;tag=as3f705421
To: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
Contact: <sip:3016@192.168.10.1:5060>
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.0
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: 269

v=0
o=root 612798355 612798356 IN IP4 192.168.10.1
s=Asterisk PBX 10.3.0
c=IN IP4 192.168.10.1
t=0 0
m=image 4369 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

---
Retransmitting #3 (NAT) to 192.168.10.2:52492:
INVITE sip:2222@192.168.10.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0e0422b7;rport
Max-Forwards: 70
From: <sip:3016@192.168.10.1>;tag=as3f705421
To: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
Contact: <sip:3016@192.168.10.1:5060>
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.0
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: 269

v=0
o=root 612798355 612798356 IN IP4 192.168.10.1
s=Asterisk PBX 10.3.0
c=IN IP4 192.168.10.1
t=0 0
m=image 4369 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

---
[Apr 26 17:06:49] WARNING[8919]: res_fax.c:1637 receivefax_t38_init: channel 'SIP/as5300-0000006c' timed-out during the T.38 negotiation.
[Apr 26 17:06:49] WARNING[8919]: res_fax.c:1687 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/as5300-0000006c' and T.38 negotiation failed; aborting.
[Apr 26 17:06:49] ERROR[8919]: res_fax.c:1891 receivefax_exec: error initializing channel 'SIP/as5300-0000006c' in T.38 mode
== Spawn extension (from-as5300, 3016, 3) exited non-zero on 'SIP/as5300-0000006c'
Scheduling destruction of SIP dialog '6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2' in 32000 ms (Method: ACK)
Retransmitting #4 (NAT) to 192.168.10.2:52492:
INVITE sip:2222@192.168.10.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0e0422b7;rport
Max-Forwards: 70
From: <sip:3016@192.168.10.1>;tag=as3f705421
To: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
Contact: <sip:3016@192.168.10.1:5060>
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.0
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: 269

v=0
o=root 612798355 612798356 IN IP4 192.168.10.1
s=Asterisk PBX 10.3.0
c=IN IP4 192.168.10.1
t=0 0
m=image 4369 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

---

<--- SIP read from UDP:192.168.10.2:57704 --->
BYE sip:3016@192.168.10.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060
From: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
To: <sip:3016@192.168.10.1>;tag=as3f705421
Date: Wed, 25 Jan 2006 19:41:46 GMT
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 6
Timestamp: 1138218109
CSeq: 102 BYE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.10.2:57704 (NAT)
Scheduling destruction of SIP dialog '6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.10.2:57704 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.2:5060;received=192.168.10.2;rport=57704
From: "Pet" <sip:2222@192.168.10.2>;tag=70DF318-21BC
To: <sip:3016@192.168.10.1>;tag=as3f705421
Call-ID: 6E14F393-8D1111DA-858F89C3-61AEE9A1@192.168.10.2
CSeq: 102 BYE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

видно, что астериск шлет реинвайт на Т38, а циска молчит. Хорошо, вот интересный дебаг со стороны циски (командой debug ccsip messages):

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Apr 26 17:11:56 as5300 21349: 1d08h: Sent:
Apr 26 17:11:56 as5300 21350: INVITE sip:3016@192.168.10.1:5060 SIP/2.0
Apr 26 17:11:56 as5300 21351: Via: SIP/2.0/UDP 192.168.10.2:5060
Apr 26 17:11:56 as5300 21352: From: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:11:56 as5300 21353: To: <sip:3016@192.168.10.1>
Apr 26 17:11:56 as5300 21354: Date: Wed, 25 Jan 2006 19:46:52 GMT
Apr 26 17:11:56 as5300 21355: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:11:56 as5300 21356: Supported: timer,100rel
Apr 26 17:11:56 as5300 21357: Min-SE: 1800
Apr 26 17:11:56 as5300 21358: Cisco-Guid: 710379384-2366771674-2240842179-1638853025
Apr 26 17:11:56 as5300 21359: User-Agent: Cisco-SIPGateway/IOS-12.x
Apr 26 17:11:56 as5300 21360: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Apr 26 17:11:56 as5300 21361: CSeq: 101 INVITE
Apr 26 17:11:56 as5300 21362: Max
Apr 26 17:11:56 as5300 21363: -Forwards: 6
Apr 26 17:11:56 as5300 21364: Remote-Party-ID: <sip:2222@192.168.10.2>;party=calling;screen=no;privacy=off
Apr 26 17:11:56 as5300 21365: Timestamp: 1138218412
Apr 26 17:11:56 as5300 21366: Contact: <sip:2222@192.168.10.2:5060>
Apr 26 17:11:56 as5300 21367: Expires: 180
Apr 26 17:11:56 as5300 21368: Allow-Events: telephone-event
Apr 26 17:11:56 as5300 21369: Content-Type: application/sdp
Apr 26 17:11:56 as5300 21370: Content-Length: 215
Apr 26 17:11:56 as5300 21371:
Apr 26 17:11:56 as5300 21372: v=0
Apr 26 17:11:56 as5300 21373: o=CiscoSystemsSIP-GW-UserAgent 6272 8219 IN IP4 192.168.10.2
Apr 26 17:11:56 as5300 21374: s=SIP Call
Apr 26 17:11:56 as5300 21375: c=IN IP4 192.168.10.2
Apr 26 17:11:56 as5300 21376: t=0 0
Apr 26 17:11:56 as5300 21377: m=audio 16528 RTP/AVP 8 19
Apr 26 17:11:56 as5300 21378: c=IN IP4 192.168.10.2
Apr 26 17:11:56 as5300 21379: a=rtpmap:8 PCMA/8000
Apr 26 17:11:56 as5300 21380: a=rtpmap:19 CN/8000
Apr 26 17:11:56 as5300 21381: a=ptime:20
Apr 26 17:11:56 as5300 21382:
Apr 26 17:11:56 as5300 21383: 1d08h: Received:
Apr 26 17:11:56 as5300 21384: SIP/2.0 100 Trying
Apr 26 17:11:56 as5300 21385: Via: SIP/2.0/UDP 192.168.10.2:5060;received=192.168.10.2;rport=55674
Apr 26 17:11:56 as5300 21386: From: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:11:56 as5300 21387: To: <sip:3016@192.168.10.1>
Apr 26 17:11:56 as5300 21388: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:11:56 as5300 21389: CSeq: 101 INVITE
Apr 26 17:11:56 as5300 21390: Server: Asterisk PBX 10.3.0
Apr 26 17:11:56 as5300 21391: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Apr 26 17:11:56 as5300 21392: Supported: replaces, timer
Apr 26 17:11:56 as5300 21393: Contact: <sip:3016@192.168.10.1:5060>
Apr 26 17:11:56 as5300 21394: Content-Length: 0
Apr 26 17:11:56 as5300 21395:
Apr 26 17:11:56 as5300 21396:
Apr 26 17:11:56 as5300 21397:
Apr 26 17:11:56 as5300 21398: 1d08h: Received:
Apr 26 17:11:56 as5300 21399: SIP/2.0 200 OK
Apr 26 17:11:56 as5300 21400: Via: SIP/2.0/UDP 192.168.10.2:5060;received=192.168.10.2;rport=55674
Apr 26 17:11:56 as5300 21401: From: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:11:56 as5300 21402: To: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:11:56 as5300 21403: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:11:56 as5300 21404: CSeq: 101 INVITE
Apr 26 17:11:56 as5300 21405: Server: Asterisk PBX 10.3.0
Apr 26 17:11:56 as5300 21406: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Apr 26 17:11:56 as5300 21407: Supported: replaces, timer
Apr 26 17:11:56 as5300 21408: Contact: <sip:3016@192.168.10.1:5060>
Apr 26 17:11:56 as5300 21409: Content-Type: application/sdp
Apr 26 17:11:56 as5300 21410: Content-Length: 177
Apr 26 17:11:56 as5300 21411:
Apr 26 17:11:56 as5300 21412: v=0
Apr 26 17:11:56 as5300 21413: o=ro
Apr 26 17:11:56 as5300 21414: ot 770445621 770445621 IN IP4 192.168.10.1
Apr 26 17:11:56 as5300 21415: s=Asterisk PBX 10.3.0
Apr 26 17:11:56 as5300 21416: c=IN IP4 192.168.10.1
Apr 26 17:11:56 as5300 21417: t=0 0
Apr 26 17:11:56 as5300 21418: m=audio 15166 RTP/AVP 8
Apr 26 17:11:56 as5300 21419: a=rtpmap:8 PCMA/8000
Apr 26 17:11:56 as5300 21420: a=ptime:20
Apr 26 17:11:56 as5300 21421: a=sendrecv
Apr 26 17:11:56 as5300 21422:
Apr 26 17:11:56 as5300 21423: 1d08h: Sent:
Apr 26 17:11:56 as5300 21424: ACK sip:3016@192.168.10.1:5060 SIP/2.0
Apr 26 17:11:56 as5300 21425: Via: SIP/2.0/UDP 192.168.10.2:5060
Apr 26 17:11:56 as5300 21426: From: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:11:56 as5300 21427: To: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:11:56 as5300 21428: Date: Wed, 25 Jan 2006 19:46:52 GMT
Apr 26 17:11:56 as5300 21429: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:11:56 as5300 21430: Max-Forwards: 6
Apr 26 17:11:56 as5300 21431: Content-Length: 0
Apr 26 17:11:56 as5300 21432: CSeq: 101 ACK
Apr 26 17:11:56 as5300 21433:
Apr 26 17:11:56 as5300 21434:
Apr 26 17:11:56 as5300 21435:
Apr 26 17:11:57 as5300 21436: 30064771072d4294967296h: %ISDN-6-CONNECT: Interface Serial1:29 is now connected to 2222 N/A
Apr 26 17:11:59 as5300 21437: 1d08h: Received:
Apr 26 17:11:59 as5300 21438: INVITE sip:2222@192.168.10.2:5060 SIP/2.0
Apr 26 17:11:59 as5300 21439: Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3f1c92b1;rport
Apr 26 17:11:59 as5300 21440: Max-Forwards: 70
Apr 26 17:11:59 as5300 21441: From: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:11:59 as5300 21442: To: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:11:59 as5300 21443: Contact: <sip:3016@192.168.10.1:5060>
Apr 26 17:11:59 as5300 21444: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:11:59 as5300 21445: CSeq: 102 INVITE
Apr 26 17:11:59 as5300 21446: User-Agent: Asterisk PBX 10.3.0
Apr 26 17:11:59 as5300 21447: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Apr 26 17:11:59 as5300 21448: Supported: replaces, timer
Apr 26 17:11:59 as5300 21449: X-asterisk-Info: SI
Apr 26 17:11:59 as5300 21450: P re-invite (External RTP bridge)
Apr 26 17:11:59 as5300 21451: Content-Type: application/sdp
Apr 26 17:11:59 as5300 21452: Content-Length: 269
Apr 26 17:11:59 as5300 21453:
Apr 26 17:11:59 as5300 21454: v=0
Apr 26 17:11:59 as5300 21455: o=root 770445621 770445622 IN IP4 192.168.10.1
Apr 26 17:11:59 as5300 21456: s=Asterisk PBX 10.3.0
Apr 26 17:11:59 as5300 21457: c=IN IP4 192.168.10.1
Apr 26 17:11:59 as5300 21458: t=0 0
Apr 26 17:11:59 as5300 21459: m=image 4952 udptl t38
Apr 26 17:11:59 as5300 21460: a=T38FaxVersion:0
Apr 26 17:11:59 as5300 21461: a=T38MaxBitRate:14400
Apr 26 17:11:59 as5300 21462: a=T38FaxRateManagement:transferredTCF
Apr 26 17:11:59 as5300 21463: a=T38FaxMaxDatagram:1400
Apr 26 17:11:59 as5300 21464: a=T38FaxUdpEC:t38UDPRedundancy
Apr 26 17:11:59 as5300 21465:
Apr 26 17:11:59 as5300 21466: 1d08h: Sent:
Apr 26 17:11:59 as5300 21467: SIP/2.0 200 OK
Apr 26 17:11:59 as5300 21468: Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3f1c92b1;rport
Apr 26 17:11:59 as5300 21469: From: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:11:59 as5300 21470: To: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:11:59 as5300 21471: Date: Wed, 25 Jan 2006 19:46:55 GMT
Apr 26 17:11:59 as5300 21472: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:11:59 as5300 21473: Server: Cisco-SIPGateway/IOS-12.x
Apr 26 17:11:59 as5300 21474: CSeq: 102 INVITE
Apr 26 17:11:59 as5300 21475: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Apr 26 17:11:59 as5300 21476: Allow-Events: telephone-event
Apr 26 17:11:59 as5300 21477: Contact: <sip:2222@192.168.10.2:5060>
Apr 26 17:11:59 as5300 21478: Content-Type: app
Apr 26 17:12:00 as5300 21479: lication/sdp
Apr 26 17:12:00 as5300 21480: Content-Length: 294
Apr 26 17:12:00 as5300 21481:
Apr 26 17:12:00 as5300 21482: v=0
Apr 26 17:12:00 as5300 21483: o=CiscoSystemsSIP-GW-UserAgent 6272 8220 IN IP4 192.168.10.2
Apr 26 17:12:00 as5300 21484: s=SIP Call
Apr 26 17:12:00 as5300 21485: c=IN IP4 192.168.10.2
Apr 26 17:12:00 as5300 21486: t=0 0
Apr 26 17:12:00 as5300 21487: m=image 16528 udptl t38
Apr 26 17:12:00 as5300 21488: c=IN IP4 192.168.10.2
Apr 26 17:12:00 as5300 21489: a=T38FaxVersion:0
Apr 26 17:12:00 as5300 21490: a=T38MaxBitRate:14400
Apr 26 17:12:00 as5300 21491: a=T38FaxRateManagement:transferredTCF
Apr 26 17:12:00 as5300 21492: a=T38FaxMaxDatagram:72
Apr 26 17:12:00 as5300 21493: a=T38FaxUdpEC:t38UDPRedundancy
Apr 26 17:12:00 as5300 21494:
Apr 26 17:12:00 as5300 21495: 1d08h: Received:
Apr 26 17:12:00 as5300 21496: INVITE sip:2222@192.168.10.2:5060 SIP/2.0
Apr 26 17:12:00 as5300 21497: Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3f1c92b1;rport
Apr 26 17:12:00 as5300 21498: Max-Forwards: 70
Apr 26 17:12:00 as5300 21499: From: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:12:00 as5300 21500: To: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:12:00 as5300 21501: Contact: <sip:3016@192.168.10.1:5060>
Apr 26 17:12:00 as5300 21502: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:12:00 as5300 21503: CSeq: 102 INVITE
Apr 26 17:12:00 as5300 21504: User-Agent: Asterisk PBX 10.3.0
Apr 26 17:12:00 as5300 21505: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Apr 26 17:12:00 as5300 21506: Supported: replaces, timer
Apr 26 17:12:00 as5300 21507: X-asterisk-Info: SI
Apr 26 17:12:00 as5300 21508: P re-invite (External RTP bridge)
Apr 26 17:12:00 as5300 21509: Content-Type: application/sdp
Apr 26 17:12:00 as5300 21510: Content-Length: 269
Apr 26 17:12:00 as5300 21511:
Apr 26 17:12:00 as5300 21512: v=0
Apr 26 17:12:00 as5300 21513: o=root 770445621 770445622 IN IP4 192.168.10.1
Apr 26 17:12:00 as5300 21514: s=Asterisk PBX 10.3.0
Apr 26 17:12:00 as5300 21515: c=IN IP4 192.168.10.1
Apr 26 17:12:00 as5300 21516: t=0 0
Apr 26 17:12:00 as5300 21517: m=image 4952 udptl t38
Apr 26 17:12:00 as5300 21518: a=T38FaxVersion:0
Apr 26 17:12:00 as5300 21519: a=T38MaxBitRate:14400
Apr 26 17:12:00 as5300 21520: a=T38FaxRateManagement:transferredTCF
Apr 26 17:12:00 as5300 21521: a=T38FaxMaxDatagram:1400
Apr 26 17:12:00 as5300 21522: a=T38FaxUdpEC:t38UDPRedundancy
Apr 26 17:12:00 as5300 21523:
Apr 26 17:12:00 as5300 21524: 1d08h: Sent:
Apr 26 17:12:00 as5300 21525: SIP/2.0 200 OK
Apr 26 17:12:00 as5300 21526: Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3f1c92b1;rport
Apr 26 17:12:00 as5300 21527: From: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:12:00 as5300 21528: To: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:12:00 as5300 21529: Date: Wed, 25 Jan 2006 19:46:55 GMT
Apr 26 17:12:00 as5300 21530: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:12:00 as5300 21531: Server: Cisco-SIPGateway/IOS-12.x
Apr 26 17:12:00 as5300 21532: CSeq: 102 INVITE
Apr 26 17:12:00 as5300 21533: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Apr 26 17:12:00 as5300 21534: Allow-Events: telephone-event
Apr 26 17:12:00 as5300 21535: Contact: <sip:2222@192.168.10.2:5060>
Apr 26 17:12:00 as5300 21536: Content-Type: app
Apr 26 17:12:00 as5300 21537: lication/sdp
Apr 26 17:12:00 as5300 21538: Content-Length: 294
Apr 26 17:12:00 as5300 21539:
Apr 26 17:12:00 as5300 21540: v=0
Apr 26 17:12:00 as5300 21541: o=CiscoSystemsSIP-GW-UserAgent 6272 8220 IN IP4 192.168.10.2
Apr 26 17:12:00 as5300 21542: s=SIP Call
Apr 26 17:12:00 as5300 21543: c=IN IP4 192.168.10.2
Apr 26 17:12:00 as5300 21544: t=0 0
Apr 26 17:12:00 as5300 21545: m=image 16528 udptl t38
Apr 26 17:12:00 as5300 21546: c=IN IP4 192.168.10.2
Apr 26 17:12:00 as5300 21547: a=T38FaxVersion:0
Apr 26 17:12:00 as5300 21548: a=T38MaxBitRate:14400
Apr 26 17:12:00 as5300 21549: a=T38FaxRateManagement:transferredTCF
Apr 26 17:12:00 as5300 21550: a=T38FaxMaxDatagram:72
Apr 26 17:12:00 as5300 21551: a=T38FaxUdpEC:t38UDPRedundancy
Apr 26 17:12:00 as5300 21552:
Apr 26 17:12:00 as5300 21553: 1d08h: Sent:
Apr 26 17:12:00 as5300 21554: SIP/2.0 200 OK
Apr 26 17:12:00 as5300 21555: Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3f1c92b1;rport
Apr 26 17:12:00 as5300 21556: From: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:12:00 as5300 21557: To: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:12:00 as5300 21558: Date: Wed, 25 Jan 2006 19:46:55 GMT
Apr 26 17:12:00 as5300 21559: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:12:00 as5300 21560: Server: Cisco-SIPGateway/IOS-12.x
Apr 26 17:12:00 as5300 21561: CSeq: 102 INVITE
Apr 26 17:12:00 as5300 21562: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Apr 26 17:12:00 as5300 21563: Allow-Events: telephone-event
Apr 26 17:12:00 as5300 21564: Contact: <sip:2222@192.168.10.2:5060>
Apr 26 17:12:00 as5300 21565: Content-Type: app
Apr 26 17:12:01 as5300 21566: lication/sdp
Apr 26 17:12:01 as5300 21567: Content-Length: 294
Apr 26 17:12:01 as5300 21568:
Apr 26 17:12:01 as5300 21569: v=0
Apr 26 17:12:01 as5300 21570: o=CiscoSystemsSIP-GW-UserAgent 6272 8220 IN IP4 192.168.10.2
Apr 26 17:12:01 as5300 21571: s=SIP Call
Apr 26 17:12:01 as5300 21572: c=IN IP4 192.168.10.2
Apr 26 17:12:01 as5300 21573: t=0 0
Apr 26 17:12:01 as5300 21574: m=image 16528 udptl t38
Apr 26 17:12:01 as5300 21575: c=IN IP4 192.168.10.2
Apr 26 17:12:01 as5300 21576: a=T38FaxVersion:0
Apr 26 17:12:01 as5300 21577: a=T38MaxBitRate:14400
Apr 26 17:12:01 as5300 21578: a=T38FaxRateManagement:transferredTCF
Apr 26 17:12:01 as5300 21579: a=T38FaxMaxDatagram:72
Apr 26 17:12:01 as5300 21580: a=T38FaxUdpEC:t38UDPRedundancy
Apr 26 17:12:01 as5300 21581:
Apr 26 17:12:01 as5300 21582: 1d08h: Received:
Apr 26 17:12:01 as5300 21583: INVITE sip:2222@192.168.10.2:5060 SIP/2.0
Apr 26 17:12:01 as5300 21584: Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3f1c92b1;rport
Apr 26 17:12:01 as5300 21585: Max-Forwards: 70
Apr 26 17:12:01 as5300 21586: From: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:12:01 as5300 21587: To: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:12:01 as5300 21588: Contact: <sip:3016@192.168.10.1:5060>
Apr 26 17:12:01 as5300 21589: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:12:01 as5300 21590: CSeq: 102 INVITE
Apr 26 17:12:01 as5300 21591: User-Agent: Asterisk PBX 10.3.0
Apr 26 17:12:01 as5300 21592: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Apr 26 17:12:01 as5300 21593: Supported: replaces, timer
Apr 26 17:12:01 as5300 21594: X-asterisk-Info: SI
Apr 26 17:12:01 as5300 21595: P re-invite (External RTP bridge)
Apr 26 17:12:01 as5300 21596: Content-Type: application/sdp
Apr 26 17:12:01 as5300 21597: Content-Length: 269
Apr 26 17:12:01 as5300 21598:
Apr 26 17:12:01 as5300 21599: v=0
Apr 26 17:12:01 as5300 21600: o=root 770445621 770445622 IN IP4 192.168.10.1
Apr 26 17:12:01 as5300 21601: s=Asterisk PBX 10.3.0
Apr 26 17:12:01 as5300 21602: c=IN IP4 192.168.10.1
Apr 26 17:12:01 as5300 21603: t=0 0
Apr 26 17:12:01 as5300 21604: m=image 4952 udptl t38
Apr 26 17:12:01 as5300 21605: a=T38FaxVersion:0
Apr 26 17:12:01 as5300 21606: a=T38MaxBitRate:14400
Apr 26 17:12:01 as5300 21607: a=T38FaxRateManagement:transferredTCF
Apr 26 17:12:01 as5300 21608: a=T38FaxMaxDatagram:1400
Apr 26 17:12:01 as5300 21609: a=T38FaxUdpEC:t38UDPRedundancy
Apr 26 17:12:01 as5300 21610:
Apr 26 17:12:01 as5300 21611: 1d08h: Sent:
Apr 26 17:12:01 as5300 21612: SIP/2.0 200 OK
Apr 26 17:12:01 as5300 21613: Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3f1c92b1;rport
Apr 26 17:12:01 as5300 21614: From: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:12:01 as5300 21615: To: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:12:01 as5300 21616: Date: Wed, 25 Jan 2006 19:46:56 GMT
Apr 26 17:12:01 as5300 21617: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:12:01 as5300 21618: Server: Cisco-SIPGateway/IOS-12.x
Apr 26 17:12:01 as5300 21619: CSeq: 102 INVITE
Apr 26 17:12:01 as5300 21620: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Apr 26 17:12:01 as5300 21621: Allow-Events: telephone-event
Apr 26 17:12:01 as5300 21622: Contact: <sip:2222@192.168.10.2:5060>
Apr 26 17:12:01 as5300 21623: Content-Type: app
Apr 26 17:12:01 as5300 21624: lication/sdp
Apr 26 17:12:01 as5300 21625: Content-Length: 294
Apr 26 17:12:01 as5300 21626:
Apr 26 17:12:01 as5300 21627: v=0
Apr 26 17:12:01 as5300 21628: o=CiscoSystemsSIP-GW-UserAgent 6272 8220 IN IP4 192.168.10.2
Apr 26 17:12:01 as5300 21629: s=SIP Call
Apr 26 17:12:01 as5300 21630: c=IN IP4 192.168.10.2
Apr 26 17:12:01 as5300 21631: t=0 0
Apr 26 17:12:01 as5300 21632: m=image 16528 udptl t38
Apr 26 17:12:01 as5300 21633: c=IN IP4 192.168.10.2
Apr 26 17:12:01 as5300 21634: a=T38FaxVersion:0
Apr 26 17:12:01 as5300 21635: a=T38MaxBitRate:14400
Apr 26 17:12:01 as5300 21636: a=T38FaxRateManagement:transferredTCF
Apr 26 17:12:01 as5300 21637: a=T38FaxMaxDatagram:72
Apr 26 17:12:01 as5300 21638: a=T38FaxUdpEC:t38UDPRedundancy
Apr 26 17:12:01 as5300 21639:
Apr 26 17:12:01 as5300 21640: 1d08h: Sent:
Apr 26 17:12:01 as5300 21641: SIP/2.0 200 OK
Apr 26 17:12:01 as5300 21642: Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3f1c92b1;rport
Apr 26 17:12:01 as5300 21643: From: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:12:01 as5300 21644: To: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:12:01 as5300 21645: Date: Wed, 25 Jan 2006 19:46:56 GMT
Apr 26 17:12:01 as5300 21646: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:12:01 as5300 21647: Server: Cisco-SIPGateway/IOS-12.x
Apr 26 17:12:01 as5300 21648: CSeq: 102 INVITE
Apr 26 17:12:01 as5300 21649: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Apr 26 17:12:01 as5300 21650: Allow-Events: telephone-event
Apr 26 17:12:01 as5300 21651: Contact: <sip:2222@192.168.10.2:5060>
Apr 26 17:12:01 as5300 21652: Content-Type: app
Apr 26 17:12:02 as5300 21653: lication/sdp
Apr 26 17:12:02 as5300 21654: Content-Length: 294
Apr 26 17:12:02 as5300 21655:
Apr 26 17:12:02 as5300 21656: v=0
Apr 26 17:12:02 as5300 21657: o=CiscoSystemsSIP-GW-UserAgent 6272 8220 IN IP4 192.168.10.2
Apr 26 17:12:02 as5300 21658: s=SIP Call
Apr 26 17:12:02 as5300 21659: c=IN IP4 192.168.10.2
Apr 26 17:12:02 as5300 21660: t=0 0
Apr 26 17:12:02 as5300 21661: m=image 16528 udptl t38
Apr 26 17:12:02 as5300 21662: c=IN IP4 192.168.10.2
Apr 26 17:12:02 as5300 21663: a=T38FaxVersion:0
Apr 26 17:12:02 as5300 21664: a=T38MaxBitRate:14400
Apr 26 17:12:02 as5300 21665: a=T38FaxRateManagement:transferredTCF
Apr 26 17:12:02 as5300 21666: a=T38FaxMaxDatagram:72
Apr 26 17:12:02 as5300 21667: a=T38FaxUdpEC:t38UDPRedundancy
Apr 26 17:12:02 as5300 21668:
Apr 26 17:12:02 as5300 21669: 1d08h: Sent:
Apr 26 17:12:02 as5300 21670: SIP/2.0 200 OK
Apr 26 17:12:02 as5300 21671: Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3f1c92b1;rport
Apr 26 17:12:02 as5300 21672: From: <sip:3016@192.168.10.1>;tag=as0b03a0bd
Apr 26 17:12:02 as5300 21673: To: "Petelin A.E." <sip:2222@192.168.10.2>;tag=712C4E8-267A
Apr 26 17:12:02 as5300 21674: Date: Wed, 25 Jan 2006 19:46:56 GMT
Apr 26 17:12:02 as5300 21675: Call-ID: 2A595C89-8D1211DA-859389C3-61AEE9A1@192.168.10.2
Apr 26 17:12:02 as5300 21676: Server: Cisco-SIPGateway/IOS-12.x
Apr 26 17:12:02 as5300 21677: CSeq: 102 INVITE
Apr 26 17:12:02 as5300 21678: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Apr 26 17:12:02 as5300 21679: Allow-Events: telephone-event
Apr 26 17:12:02 as5300 21680: Contact: <sip:2222@192.168.10.2:5060>
Apr 26 17:12:02 as5300 21681: Content-Type: app
Apr 26 17:12:02 as5300 21682: lication/sdp
Apr 26 17:12:02 as5300 21683: Content-Length: 294
Apr 26 17:12:02 as5300 21684:
Apr 26 17:12:02 as5300 21685: v=0
Apr 26 17:12:02 as5300 21686: o=CiscoSystemsSIP-GW-UserAgent 6272 8220 IN IP4 192.168.10.2
Apr 26 17:12:02 as5300 21687: s=SIP Call
Apr 26 17:12:02 as5300 21688: c=IN IP4 192.168.10.2
Apr 26 17:12:02 as5300 21689: t=0 0
Apr 26 17:12:02 as5300 21690: m=image 16528 udptl t38
Apr 26 17:12:02 as5300 21691: c=IN IP4 192.168.10.2
Apr 26 17:12:02 as5300 21692: a=T38FaxVersion:0
Apr 26 17:12:02 as5300 21693: a=T38MaxBitRate:14400
Apr 26 17:12:02 as5300 21694: a=T38FaxRateManagement:transferredTCF
Apr 26 17:12:02 as5300 21695: a=T38FaxMaxDatagram:72
Apr 26 17:12:02 as5300 21696: a=T38FaxUdpEC:t38UDPRedundancy

т.е. видно что на каждый инвайт Т38 астериска циска ответила ОК.
Вот еще - это фрагмент вывода tcpdump захваченных пакетов на машине где стоит астериск:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

17:17:51.709077 IP 192.168.10.1.5060 > 192.168.10.2.52925: SIP, length: 869
E.......@.8j..
...
......m..INVITE sip:2222@192.168.10.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3ec2c183;rport
Max-Forwards: 70
From: <sip:3016@192.168.10.1>;tag=as41c0aac4
To: "Pet" <sip:2222@192.168.10.2>;tag=7181D64-E5E
Contact: <sip:3016@192.168.10.1:5060>
Call-ID: FB292A4B-8D1211DA-859789C3-61AEE9A1@192.168.10.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.0
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: 269

v=0
o=root 446174770 446174771 IN IP4 192.168.10.1
s=Asterisk PBX 10.3.0
c=IN IP4 192.168.10.1
t=0 0
m=image 4778 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

17:17:51.716784 IP 192.168.10.2.5060 > 192.168.10.1.5060: SIP, length: 841
Eh.e..........
...
......Q{.SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK3ec2c183;rport
From: <sip:3016@192.168.10.1>;tag=as41c0aac4
To: "Pet" <sip:2222@192.168.10.2>;tag=7181D64-E5E
Date: Wed, 25 Jan 2006 19:52:47 GMT
Call-ID: FB292A4B-8D1211DA-859789C3-61AEE9A1@192.168.10.2
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:2222@192.168.10.2:5060>
Content-Type: application/sdp
Content-Length: 293

v=0
o=CiscoSystemsSIP-GW-UserAgent 7312 234 IN IP4 192.168.10.2
s=SIP Call
c=IN IP4 192.168.10.2
t=0 0
m=image 19280 udptl t38
c=IN IP4 192.168.10.2
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy

17:17:51.718404 IP 192.168.10.2.54947 > 192.168.10.1.syslog: SYSLOG local6.debug, length: 24
E..4V.........
...
...... ..<183>22243: ication/sdp
17:17:51.718574 IP 192.168.10.2.54947 > 192.168.10.1.syslog: SYSLOG local6.debug, length: 32

т.е. видно что ответы циски до машины дошли, а до астериска нет.. Эту задачу я пока решить не могу.. подскажите куда копать?
конфиг as5300:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

version 12.3
service timestamps debug uptime
service timestamps log uptime
service password-encryption
!
hostname Cisco5300
!
boot-start-marker
boot system flash c5300-js-mz.123-26.bin
boot-end-marker
!
!
!
!
resource-pool disable
!
aaa new-model
!
!
aaa authentication login default local
aaa session-id common
ip subnet-zero
!
!
isdn switch-type primary-net5
!
!
voice service voip
sip
!
!
!
!
!
!
!
!
!
!
!
controller E1 0
shutdown
channel-group 0 timeslots 1-31
!
controller E1 1
framing NO-CRC4
clock source line primary
pri-group timeslots 1-31
description ISDN PRI link to Definity
!
controller E1 2
framing NO-CRC4
description ISDN PRI link to Definity
!
controller E1 3
channel-group 0 timeslots 1-31
!
!
!
interface Ethernet0
no ip address
shutdown
!
interface Serial0:0
no ip address
shutdown
!
interface Serial1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice modem
no cdp enable
!
interface Serial3:0
no ip address
shutdown
!
interface FastEthernet0
description LINK-TO-PLUTO
ip address 192.168.10.2 255.255.255.252
duplex auto
speed auto
h323-gateway voip interface
!
ip classless
ip route 0.0.0.0 0.0.0.0 192.168.10.1
no ip http server
!
!
logging trap debugging
logging facility local6
logging source-interface FastEthernet0
logging 192.168.10.1
!
!
!
!
voice-port 1:D
cptone RU
!
!
!
dial-peer voice 1 voip
huntstop
incoming called-number .
destination-pattern .%
session protocol sipv2
session target ipv4:192.168.10.1
codec g711alaw
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
!
dial-peer voice 100 pots
description Outgoing-to-ISDN
destination-pattern 9
direct-inward-dial
port 1:D
!
sip-ua
sip-server ipv4:192.168.10.1
!
!
line con 0
line aux 0
line vty 0 4
session-timeout 60
exec-timeout 300 0
!
end

sip.conf астериска:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[general]
context=default
allowguest=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=no
engine=asterisk
t38pt_udptl = yes,redundancy
faxdetect = t38

[1351]
type=friend
host=dynamic
username=1351
secret=blabla
nat=no
canreinvite=no
context=default
callerid=”Test1” <1351>
disallow=all
allow=g729
allow=allaw

[1350]
type=friend
host=dynamic
username=1350
secret=blabla
nat=no
canreinvite=no
context=default
callerid=”Test2” <1350>
disallow=all
allow=g729
allow=alaw



[as5300]
host=192.168.10.2
type=friend
insecure=port,invite
context=from-as5300
disallow=all
allow=alaw
allow=g729

[cisco]
host=192.168.10.2
type=friend
;insecure=port,invite
context=to-as5300
disallow=all
allow=alaw
allow=g729

extensions.conf

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp
TRUNK=DAHDI/G2
TRUNKMSD=1

[default]
include => from-internal
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,BackGround(local-welcomе)
exten => s,4,WaitExten
exten => 1350,1,Dial(SIP/${EXTEN},30,t)
exten => 1351,1,Dial(SIP/${EXTEN},30,t)
exten => _3016,1,Answer
exten => _3016,n,Set(DT=${UNIQUEID})
exten => _3016,n,ReceiveFAX(/var/spool/asterisk/fax/${DT}.tif)
exten => _3016,n,Wait(1)
exten => 5353,1,VoiceMailMain
exten => 5353,2,Hangup

[from-as5300]
include => default

[to-as5300]
exten => _9.,1,Dial(SIP/${EXTEN}@cisco,30,t)

[ded]

Вот еще - это фрагмент вывода tcpdump захваченных пакетов на машине где стоит астериск:

Код: Выделить всё

CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.0

т.е. видно что ответы циски до машины дошли, а до астериска нет.

Я вижу, что до астериска дошли как раз. Он же ответил - User-Agent: Asterisk PBX 10.3.0?
Частное мнение: результативность решения траблов по факсам на сложных схемах, включающих t38,sip---- as5300 ---E1--- Avaya Definity не стоит затраченых Ваших усилий, наших усилий, никаких и ничьих усилий. Платим золотой монетой за деревянный товар.
http://asterisk.ru/knowledgebase/foip

[Psyhoed]

Уважаемый ded, да, читал о foip, понимаю, что факсы не обязаны ходить поверх ip. Имею некоторое представление о том, что такое потери пакетов, джитер, ЦАП-АЦП, сжатие и вносимые ими искажения и тд... так же отдаю себе отчет, что напрягаюсь сам, Вас и вряд ли это кто то оценит пока это все будет работать, зато когда что то упадет, отвалится или просто перестанет работать сразу вспомнят как обычно не хорошим словом
Но.. помните как в книге В. Каверина "Два капитана" - "Бороться, искать, найти и не сдаваться" (за точность цитаты не ручаюсь )
На счет вывода пакетов tcpdump:

17:17:51.709077 IP 192.168.10.1.5060 > 192.168.10.2.52925: SIP, length: 869
E.......@.8j..
...
......m..INVITE sip:2222@192.168.10.2:5060 SIP/2.0

192.168.10.1 - это астериск шлет as5300 реинвайт на Т38, на что as5300 (192.168.10.2) отвечает

17:17:51.716784 IP 192.168.10.2.5060 > 192.168.10.1.5060: SIP, length: 841
Eh.e..........
...
......Q{.SIP/2.0 200 OK

НО! в тоже время в CLI астериска я вижу только:

INVITE sip:2222@192.168.10.2:5060 SIP/2.0
....
Retransmitting #1 (NAT) to 192.168.10.2:52492:
INVITE sip:2222@192.168.10.2:5060 SIP/2.0
....
Retransmitting #2 (NAT) to 192.168.10.2:52492:
INVITE sip:2222@192.168.10.2:5060 SIP/2.0
....

вот собственно и родился вопрос - Куда делся ОК от циски, если tcpdump на этой же машине его ловит, а астериск нет?
да, можно подумать о файрволе, проблемах с роутами, но голосовые вызовы то ходят на ура..
у меня приходят мысли или я еще чего то не знаю о конфигурации астериска (что скорее всего, т.к. "дружу" с ним не так давно) или.. т.к. операционка на машине где стоит астериск Gentoo Linux, собиралось все из исходников, asterisk, spandsp.. возможно забыл включить поддержку чего то при компиляции.. но это я завтра проверю.. отпишусь.

[ded]

Раз Астериск ретрансмитит по нескольку раз (и как бы не замечает ОК), то этот ответ его чем-то не устраивает. Не могу я лезь в RFC прверять порядок обмена.
По пробуйте нати хитрый параметр pedantic=yes в конфигурации sip.conf http://forum.asterisk.ru/viewtopic.php?f=3&t=443

"Бороться, искать, найти и не сдаваться" - это девиз Роберта Скотта, который всё-таки дошёл до Южного полюса, и увидел, что норвежцы уже поставили там свой флаг. сама фраза - из поэмы У. Тениссона "Уллис" (за точность цитаты ручаюсь). Каверин позаимсвовал её в "Двух капитанах" конечно, в лучшем смысле этого слова.
Ну так вот он заплатил за этот девиз своей жизнью и жизнью своих товарищей - на обратном пути замёрзли. Готовы ли Вы не жизнью конечно, а своими нейронами и спинномозговой жидкостью жертвовать на то, чтобы какие-то офисные инфузории отсылали бумагу глупым казённым образом?

[Psyhoed]

Каждый сам выбирает путь.. разве цена установки флага (а тем более не установки) должна быть человеческие жизни? они знали на что идут (по крайней мере догадывались). Так и здесь.. основная масса пользователей сетями связи - не связисты.. А мы выбрали путь создания/обслуживания этих сетей

Уважаемый ded, большое спасибо, что откликнулись. Забегал этот проклятый Т38 везде. И проблема была совсем не в астериске.
Kernel net filter connection tracking sip module. Вот виновник. Если бы я сразу посмотрел лог ядра, то и не приставал бы тут с глупыми вопросами
а там увидел следующее:

nf_ct_sip: dropping packetIN=eth0 OUT= MAC=00:04:79:67:1e:88:00:e0:1e75:45:08:00 SRC=192.168.10.2 DST=192.168.10.1 LEN=861 TOS
=0x08 PREC=0x60 TTL=255 ID=1798 PROTO=UDP SPT=5060 DPT=5060 LEN=841

после выгрузки этого модуля все забегало. Непонятно почему они дропал только ответы на Т38 инвайт и только на циске, а на другой сетевухе все пропускал... но это уже другая тема

[ded]

Kernel net filter connection tracking sip module? Даже не слышал никогда про такой, и не подозревал что такой может быть! Уточните версию и дистрибутив Gentoo?
Ваше сообщение - это ценный вклад в копилку. Такой тупик найти - невообразимо!

[Psyhoed]

Вот с версией Gentoo сложнее, это один из current релизов, примерно февраля этого года. Ядро версии 3.3.1.
вот вывод uname -a:

Linux localhost 3.3.1-gentoo #3 SMP Fri Apr 27 11:07:05 MSK 2012 i686 Intel(R) Xeon(TM) CPU 2.66GHz GenuineIntel GNU/Linux

kernel net filter connection tracking sip module - я имел ввиду nf_conntrack_sip.ko собранный модулем, а не включенный в ядро монолитно. Строка в .config CONFIG_NF_CONNTRACK_SIP=m
короче, это модуль iptables
из описания:

SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions (conferences) such as Internet telephony calls. With the ip_conntrack_sip and the nf_nat_sip modules you can support the protocol on a connection tracking/NATing firewall

Непонятно, почему без созданных в iptables политик и цепочек он дропал пакеты.
И не надо преувеличивать мой вклад, миллионы людей ежедневно пользуют линукс и астериск, думаю уже кто либо сталкивался))

[jugatsu]

Да ладно, пр INPUT ACCEPT дропал пакеты? Не верю Покажи iptables-save

[Psyhoed]

Вы абсолютно правы с INPUT ACCEPT дропает.

# lsmod
Module Size Used by
nf_conntrack_sip 15440 0

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

# iptables-save
# Generated by iptables-save v1.4.13 on Fri Apr 27 15:14:34 2012
*security
:INPUT ACCEPT [24161:1919933]
:FORWARD ACCEPT [0:0]
:OUTPUT ACCEPT [21191:4037641]
COMMIT
# Completed on Fri Apr 27 15:14:34 2012
# Generated by iptables-save v1.4.13 on Fri Apr 27 15:14:34 2012
*raw
:PREROUTING ACCEPT [36527:3627202]
:OUTPUT ACCEPT [21192:4037837]
COMMIT
# Completed on Fri Apr 27 15:14:34 2012
# Generated by iptables-save v1.4.13 on Fri Apr 27 15:14:34 2012
*nat
:PREROUTING ACCEPT [15:3553]
:INPUT ACCEPT [15:3553]
:OUTPUT ACCEPT [1:467]
:POSTROUTING ACCEPT [1:467]
COMMIT
# Completed on Fri Apr 27 15:14:34 2012
# Generated by iptables-save v1.4.13 on Fri Apr 27 15:14:34 2012
*mangle
:PREROUTING ACCEPT [36527:3627202]
:INPUT ACCEPT [26264:2426053]
:FORWARD ACCEPT [0:0]
:OUTPUT ACCEPT [21197:4038737]
:POSTROUTING ACCEPT [21197:4038737]
COMMIT
# Completed on Fri Apr 27 15:14:34 2012
# Generated by iptables-save v1.4.13 on Fri Apr 27 15:14:34 2012
*filter
:INPUT ACCEPT [3124:217752]
:FORWARD ACCEPT [0:0]
:OUTPUT ACCEPT [2352:323729]
COMMIT
# Completed on Fri Apr 27 15:14:34 2012

в messages:

Apr 27 15:12:17 localhost kernel: [14017.566908] nf_ct_sip: dropping packetIN=eth0 OUT= MAC=00:04:79:67:1e:88:00:e0:1e75:45:08:00 SRC=192.168.10.2 DST=192.168.10.1 LEN=864 TOS=
0x08 PREC=0x60 TTL=255 ID=1875 PROTO=UDP SPT=5060 DPT=5060 LEN=844
Apr 27 15:12:18 localhost kernel: [14018.063873] nf_ct_sip: dropping packetIN=eth0 OUT= MAC=00:04:79:67:1e:88:00:e0:1e75:45:08:00 SRC=192.168.10.2 DST=192.168.10.1 LEN=864 TOS=
0x08 PREC=0x60 TTL=255 ID=1876 PROTO=UDP SPT=5060 DPT=5060 LEN=844

как только делаешь rmmod nf_conntrack_sip - все бегает..

[jugatsu]

Ну явно бага же, такого не должно быть, я полагаю А чем обусловлен выбор Генту, если не серет * не холивара ради *