При параметрах insecure=invite,port еще веселей, разрывает связь через 14 сек.

Тогда sip debug и в путь, разбираться что куда отправляется и откуда принимается.

Ниже debug по пиру sipnet. Далее 6 Retransmitting и обрыв.
Сейчас еще tcpdump посмотрю, может чего интересного покажет.

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2010.03.09 21:26:27 =~=~=~=~=~=~=~=~=~=~=~=


ccall*CLI>sip set debug peer sipnet

ccall*CLI>
SIP Debugging Enabled for IP: 212.53.40.40:5060

ccall*CLI>

<--- SIP read from UDP://212.53.40.40:5060 --->
INVITE sip:SIP-ID-USER@IP.ASTERISK SIP/2.0

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK788128-kmbdctk;cgp=etc.tario.ru;upaddr=81.88.80.36;rport

P-CGP-Redirector: SIP-ID@sipnet.ru

Record-Route: <sip:212.53.40.40:5060;lr>

Record-Route: <sip:192.168.40.72:5060;lr>

Record-Route: <sip:rev.672236-192.168.40.72.dialog.cgatepro;lr>

Via: SIP/2.0/UDP 81.88.80.36;branch=z9hG4bK928a.f6e719433010a1eb0dc2953ec71bd7c9.0;i=4

Via: SIP/2.0/TCP 192.168.30.19:5060;branch=z9hG4bK-d8754z-b5c8802667b4156c-1---d8754z-;rport=33068

Max-Forwards: 15

From: <sip:7495-MANGO-NOMER@mangosip.ru>;tag=40f2c64e

To: <sip:SIP-ID-USER@sipnet.ru>

Call-ID: ZTE0Y2VmM2ViNTJkMThkNmRhODk0MDhkZGIyZjE5YzU.

Contact: <sip:7495-MANGO-NOMER@mangosip.ru>

CSeq: 2 INVITE

User-Agent: Softswitch2

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REGISTER

Content-Type: application/sdp

RtpProxy1: force

Content-Length: 209



v=0

o=- 167635408 167635408 IN IP4 192.168.11.240

s=-

c=IN IP4 81.88.80.36


ccall*CLI>
t=0 0

m=audio 8276 RTP/AVP 8 96

c=IN IP4 81.88.80.36

a=rtpmap:8 PCMA/8000

a=rtpmap:96 telephone-event/8000

a=nortpproxy:yes


<------------->
--- (19 headers 10 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 212.53.40.40 : 5060 (NAT)
Using INVITE request as basis request - ZTE0Y2VmM2ViNTJkMThkNmRhODk0MDhkZGIyZjE5YzU.
No user '7495-MANGO-NOMER' in SIP users list
Found peer 'sipnet' for '7495-MANGO-NOMER' from 212.53.40.40:5060
Found RTP audio format 8
Found RTP audio format 96
Peer audio RTP is at port 81.88.80.36:8276
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 96
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 81.88.80.36:8276
Looking for SIP-ID-USER in test (domain IP.ASTERISK)
list_route: hop: <sip:212.53.40.40:5060;lr>
list_route: hop: <sip:192.168.40.72:5060;lr>
list_route: hop: <sip:rev.672236-192.168.40.72.dialog.cgatepro;lr>

<--- Transmitting (NAT) to 212.53.40.40:5060 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK788128-kmbdctk;cgp=etc.tario.ru;upaddr=81.88.80.36;received=212.53.40.40;rport=5060

Via: SIP/2.0/UDP 81.88.80.36;branch=z9hG4bK928a.f6e719433010a1eb0dc2953ec71bd7c9.0;i=4

Via: SIP/2.0/TCP 192.168.30.19:5060;branch=z9hG4bK-d8754z-b5c8802667b4156c-1---d8754z-;rport=33068

Record-Route: <sip:212.53.40.40:5060;lr>

Record-Route: <sip:192.168.40.72:5060;lr>

Record-Route: <sip:rev.672236-192.168.40.72.dialog.cgatepro;lr>

From: <sip:7495-MANGO-NOMER@mangosip.ru>;tag=40f2c64e

To: <sip:SIP-ID-USER@sipnet.ru>

Call-ID: ZTE0Y2VmM2ViNTJkMThkNmRhODk0MDhkZGIyZjE5YzU.

CSeq: 2 INVITE

User-Agent: dlink 12-38-28928749-0.9.5.1.1140-IAD20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:SIP-ID-USER@IP.ASTERISK>

Content-Length: 0




<------------>
-- Executing [SIP-ID-USER@test:1] Dial("SIP/SIP-ID-USER-083c0008", "SIP/100101,120") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 100101

ccall*CLI>
-- SIP/100101-08478ed8 is ringing

<--- Transmitting (NAT) to 212.53.40.40:5060 --->
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK788128-kmbdctk;cgp=etc.tario.ru;upaddr=81.88.80.36;received=212.53.40.40;rport=5060

Via: SIP/2.0/UDP 81.88.80.36;branch=z9hG4bK928a.f6e719433010a1eb0dc2953ec71bd7c9.0;i=4

Via: SIP/2.0/TCP 192.168.30.19:5060;branch=z9hG4bK-d8754z-b5c8802667b4156c-1---d8754z-;rport=33068

Record-Route: <sip:212.53.40.40:5060;lr>

Record-Route: <sip:192.168.40.72:5060;lr>

Record-Route: <sip:rev.672236-192.168.40.72.dialog.cgatepro;lr>

From: <sip:7495-MANGO-NOMER@mangosip.ru>;tag=40f2c64e

To: <sip:SIP-ID-USER@sipnet.ru>;tag=as1b8cd385

Call-ID: ZTE0Y2VmM2ViNTJkMThkNmRhODk0MDhkZGIyZjE5YzU.

CSeq: 2 INVITE

User-Agent: dlink 12-38-28928749-0.9.5.1.1140-IAD20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:SIP-ID-USER@IP.ASTERISK>

Content-Length: 0




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

ccall*CLI>
-- SIP/100101-08478ed8 answered SIP/SIP-ID-USER-083c0008
Audio is at IP.ASTERISK port 14694
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 212.53.40.40:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK788128-kmbdctk;cgp=etc.tario.ru;upaddr=81.88.80.36;received=212.53.40.40;rport=5060

Via: SIP/2.0/UDP 81.88.80.36;branch=z9hG4bK928a.f6e719433010a1eb0dc2953ec71bd7c9.0;i=4

Via: SIP/2.0/TCP 192.168.30.19:5060;branch=z9hG4bK-d8754z-b5c8802667b4156c-1---d8754z-;rport=33068

Record-Route: <sip:212.53.40.40:5060;lr>

Record-Route: <sip:192.168.40.72:5060;lr>

Record-Route: <sip:rev.672236-192.168.40.72.dialog.cgatepro;lr>

From: <sip:7495-MANGO-NOMER@mangosip.ru>;tag=40f2c64e

To: <sip:SIP-ID-USER@sipnet.ru>;tag=as1b8cd385

Call-ID: ZTE0Y2VmM2ViNTJkMThkNmRhODk0MDhkZGIyZjE5YzU.

CSeq: 2 INVITE

User-Agent: dlink 12-38-28928749-0.9.5.1.1140-IAD20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:SIP-ID-USER@IP.ASTERISK>

Content-Type: application/sdp

Content-Length: 258



v=0

o=root 1093903183 1093903183 IN IP4 IP.ASTERISK

s=Asterisk PBX 1.6.0.9

c=IN IP4 IP.ASTERISK

t=0 0

m=audio 14694 RTP/AVP 8 96

a=rtpmap:8 PCMA/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


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

ccall*CLI>
Retransmitting #1 (NAT) to 212.53.40.40:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK788128-kmbdctk;cgp=etc.tario.ru;upaddr=81.88.80.36;received=212.53.40.40;rport=5060

Via: SIP/2.0/UDP 81.88.80.36;branch=z9hG4bK928a.f6e719433010a1eb0dc2953ec71bd7c9.0;i=4

Via: SIP/2.0/TCP 192.168.30.19:5060;branch=z9hG4bK-d8754z-b5c8802667b4156c-1---d8754z-;rport=33068

Record-Route: <sip:212.53.40.40:5060;lr>

Record-Route: <sip:192.168.40.72:5060;lr>

Record-Route: <sip:rev.672236-192.168.40.72.dialog.cgatepro;lr>

From: <sip:7495-MANGO-NOMER@mangosip.ru>;tag=40f2c64e

To: <sip:SIP-ID-USER@sipnet.ru>;tag=as1b8cd385

Call-ID: ZTE0Y2VmM2ViNTJkMThkNmRhODk0MDhkZGIyZjE5YzU.

CSeq: 2 INVITE

User-Agent: dlink 12-38-28928749-0.9.5.1.1140-IAD20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:SIP-ID-USER@IP.ASTERISK>

Content-Type: application/sdp

Content-Length: 258



v=0

o=root 1093903183 1093903183 IN IP4 IP.ASTERISK

s=Asterisk PBX 1.6.0.9

c=IN IP4 IP.ASTERISK

t=0 0

m=audio 14694 RTP/AVP 8 96

a=rtpmap:8 PCMA/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


---

а точно в параметрах xlite не стоит "In times of network disruption, automatically hang up calls after"? (advenced ->network)
Если включено, отключить и будет счастье

Ваш dlink похоже не настроен на canreinvite=no и пытается отвечать на INVITE напрямую в sipnet.

sip showp peer dlink
у него nat=? canreinvite=?

это не D-link - Это Asterisk (User-Agent прописан как D-Link). nat=yes canreinvite=no

"In times of network disruption, automatically hang up calls after" - это уже убрано. Тут все Ок.

Кстати на другом астериске (Trixbox 2.8) тоже такая же проблема, когда прописываешь этот транк. Правда там * тоже за NAT стоит, но у меня статик IP, а там вообще динамический адрес.

Вам это уже не поможет. Нужно делать полный sip debug.

Попробовал на Asterisk сделать ответ на звонок и проиграть музыку, музыка играет 20 сек. и разрыв. Получается что дело точно не клиенте, а на стыке SIPNET <-> Asterisk.

SIP-ID-USER – Это ID в SIPNET
7495-MANGO-NOMER – Московский номер МАНГО на который приходит звонок, дальше МАНГО перекидывает на SIPNET.
192.168.205.X – Моя внутренняя подсеть
EXT.IP.ASTERISK – Внешний адрес для Астериска
User-Agent: dlink - Это Астериск
192.168.205.33 – Внутренний адрес Астериска (Интерфейс один)
192.168.205.1 – Клиент X-Lite 1001 – куда приходит входящий звонок.


Если включить sip debug то получим, Retransmitting 1 по 5 удалил, они все как 6 в логе ниже.

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2010.03.09 22:26:26 =~=~=~=~=~=~=~=~=~=~=~=
ccall*CLI>
SIP Debugging enabled

------------->

<--- SIP read from UDP://212.53.40.40:5060 --->
INVITE sip:SIP-ID-USER@EXT.IP.ASTERISK SIP/2.0

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK833518-kmbdctj;cgp=etc.tario.ru;upaddr=81.88.80.36;rport

P-CGP-Redirector: SIP-ID-USER@sipnet.ru

Record-Route: <sip:212.53.40.40:5060;lr>

Record-Route: <sip:192.168.40.71:5060;lr>

Record-Route: <sip:rev.684527-192.168.40.71.dialog.cgatepro;lr>

Via: SIP/2.0/UDP 81.88.80.36;branch=z9hG4bKa6fe.e80a14020be51ff70c83500ef2acf327.0;i=4

Via: SIP/2.0/TCP 192.168.30.19:5060;branch=z9hG4bK-d8754z-60a0fc090951fa7a-1---d8754z-;rport=33068

Max-Forwards: 15

From: <sip:7495-MANGO-NOMER@mangosip.ru>;tag=d4456a6d

To: <sip:SIP-ID-USER@sipnet.ru>

Call-ID: YmIyNjE2NTM5ODBjYzJmZjJlZDI4YjMzMzA5MTdkMzY.

Contact: <sip:7495-MANGO-NOMER@mangosip.ru>

CSeq: 2 INVITE

User-Agent: Softswitch2

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REGISTER

Content-Type: application/sdp

RtpProxy1: force

Content-Length: 209


v=0

o=- 195515408 195515408 IN IP4 192.168.11.240

s=-

c=IN IP4 81.88.80.36

t=0 0

m=audio 8602 RTP/AVP 8 96

c=IN IP4 81.88.80.36

a=rtpmap:8 PCMA/8000

a=rtpmap:96 telephone-event/8000

a=nortpproxy:yes


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

[ccall*CLI>
--- (19 headers 10 lines) ---
[ccall*CLI>
== Using SIP RTP TOS bits 184
[Kccall*CLI>
== Using SIP RTP CoS mark 5
[ccall*CLI>
== Using SIP VRTP TOS bits 136
[ccall*CLI>
== Using SIP VRTP CoS mark 6
[ccall*CLI>
== Using UDPTL TOS bits 184
[call*CLI>
== Using UDPTL CoS mark 5
[ccall*CLI>
Sending to 212.53.40.40 : 5060 (NAT)
[Kccall*CLI>
Using INVITE request as basis request - YmIyNjE2NTM5ODBjYzJmZjJlZDI4YjMzMzA5MTdkMzY.
[Kccall*CLI>
No user '7495-MANGO-NOMER' in SIP users list
[Kccall*CLI>
Found peer 'sipnet' for '7495-MANGO-NOMER' from 212.53.40.40:5060
[Kccall*CLI>
Found RTP audio format 8
[Kccall*CLI>
Found RTP audio format 96
[Kccall*CLI>
Peer audio RTP is at port 81.88.80.36:8602
[Kccall*CLI>
Found audio description format PCMA for ID 8
[Kccall*CLI>
Found audio description format telephone-event for ID 96
[Kccall*CLI>
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Kccall*CLI>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Kccall*CLI>
Peer audio RTP is at port 81.88.80.36:8602
[Kccall*CLI>
Looking for SIP-ID-USER in test (domain EXT.IP.ASTERISK)
[Kccall*CLI>
list_route: hop: <sip:212.53.40.40:5060;lr>
[Kccall*CLI>
list_route: hop: <sip:192.168.40.71:5060;lr>
[Kccall*CLI>
list_route: hop: <sip:rev.684527-192.168.40.71.dialog.cgatepro;lr>
[Kccall*CLI>

<--- Transmitting (NAT) to 212.53.40.40:5060 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK833518-kmbdctj;cgp=etc.tario.ru;upaddr=81.88.80.36;received=212.53.40.40;rport=5060

Via: SIP/2.0/UDP 81.88.80.36;branch=z9hG4bKa6fe.e80a14020be51ff70c83500ef2acf327.0;i=4

Via: SIP/2.0/TCP 192.168.30.19:5060;branch=z9hG4bK-d8754z-60a0fc090951fa7a-1---d8754z-;rport=33068

Record-Route: <sip:212.53.40.40:5060;lr>

Record-Route: <sip:192.168.40.71:5060;lr>

Record-Route: <sip:rev.684527-192.168.40.71.dialog.cgatepro;lr>

From: <sip:7495-MANGO-NOMER@mangosip.ru>;tag=d4456a6d

To: <sip:SIP-ID-USER@sipnet.ru>

Call-ID: YmIyNjE2NTM5ODBjYzJmZjJlZDI4YjMzMzA5MTdkMzY.

CSeq: 2 INVITE

User-Agent: dlink 12-38-28928749-0.9.5.1.1140-IAD20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:SIP-ID-USER@EXT.IP.ASTERISK>

Content-Length: 0




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

[Kccall*CLI>
-- Executing [SIP-ID-USER@test:1] [1;36;40mDial[0;37;40m("[1;35;40mSIP/SIP-ID-USER-083c0008[0;37;40m", "[1;35;40mSIP/1001,120[0;37;40m") in new stack

[Kccall*CLI>
== Using SIP RTP TOS bits 184

[Kccall*CLI>
== Using SIP RTP CoS mark 5
[Kccall*CLI>
== Using SIP VRTP TOS bits 136
[Kccall*CLI>
== Using SIP VRTP CoS mark 6
[Kccall*CLI>
== Using UDPTL TOS bits 184
[Kccall*CLI>
== Using UDPTL CoS mark 5
[Kccall*CLI>
Audio is at 192.168.205.33 port 10446
[Kccall*CLI>
Video is at 192.168.205.33 port 16962
[Kccall*CLI>
Adding codec 0x8 (alaw) to SDP
[Kccall*CLI>
Adding codec 0x4 (ulaw) to SDP
[Kccall*CLI>
Adding codec 0x100 (g729) to SDP
[Kccall*CLI>
Adding codec 0x1 (g723) to SDP
[Kccall*CLI>
Adding codec 0x2 (gsm) to SDP
[Kccall*CLI>
Adding video codec 0x80000 (h263) to SDP
[Kccall*CLI>
Adding video codec 0x100000 (h263p) to SDP
[Kccall*CLI>
Adding video codec 0x200000 (h264) to SDP
[Kccall*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kccall*CLI>
Reliably Transmitting (NAT) to 192.168.205.1:28874:
INVITE sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef SIP/2.0

Via: SIP/2.0/UDP 192.168.205.33:5060;branch=z9hG4bK5220e964;rport

Max-Forwards: 70

From: "7495-MANGO-NOMER" <sip:7495-MANGO-NOMER@192.168.205.33>;tag=as7d591f30

To: <sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef>

Contact: <sip:7495-MANGO-NOMER@192.168.205.33>

Call-ID: 522ff9010b926a495491d3be2dc6e8b6@192.168.205.33

CSeq: 102 INVITE

User-Agent: dlink 12-38-28928749-0.9.5.1.1140-IAD20

Date: Tue, 09 Mar 2010 19:26:53 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 534



v=0

o=root 385695083 385695083 IN IP4 192.168.205.33

s=Asterisk PBX 1.6.0.9

c=IN IP4 192.168.205.33

b=CT:384

t=0 0

m=audio 10446 RTP/AVP 8 0 18 4 3 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:4 G723/8000

a=fmtp:4 annexa=no

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

m=video 16962 RTP/AVP 34 98 99

a=rtpmap:34 H263/90000

a=rtpmap:98 h263-1998/90000

a=rtpmap:99 H264/90000

a=sendrecv


---

[Kccall*CLI>
-- Called 1001

[Kccall*CLI>

<--- SIP read from UDP://192.168.205.1:28874 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.205.33:5060;branch=z9hG4bK5220e964;rport=5060

To: <sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef>

From: "7495-MANGO-NOMER" <sip:7495-MANGO-NOMER@192.168.205.33>;tag=as7d591f30

Call-ID: 522ff9010b926a495491d3be2dc6e8b6@192.168.205.33

CSeq: 102 INVITE

Content-Length: 0

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

[Kccall*CLI>
--- (7 headers 0 lines) ---

[Kccall*CLI>

<--- SIP read from UDP://192.168.205.1:28874 --->
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.205.33:5060;branch=z9hG4bK5220e964;rport=5060

Contact: <sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef>

To: <sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef>;tag=96462061

From: "7495-MANGO-NOMER"<sip:7495-MANGO-NOMER@192.168.205.33>;tag=as7d591f30

Call-ID: 522ff9010b926a495491d3be2dc6e8b6@192.168.205.33

CSeq: 102 INVITE

User-Agent: eyeBeam release 1102u stamp 52345

Content-Length: 0

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

[Kccall*CLI>
--- (9 headers 0 lines) ---

[Kccall*CLI>
-- SIP/1001-08478ed8 is ringing

[Kccall*CLI>

<--- Transmitting (NAT) to 212.53.40.40:5060 --->
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK833518-kmbdctj;cgp=etc.tario.ru;upaddr=81.88.80.36;received=212.53.40.40;rport=5060

Via: SIP/2.0/UDP 81.88.80.36;branch=z9hG4bKa6fe.e80a14020be51ff70c83500ef2acf327.0;i=4

Via: SIP/2.0/TCP 192.168.30.19:5060;branch=z9hG4bK-d8754z-60a0fc090951fa7a-1---d8754z-;rport=33068

Record-Route: <sip:212.53.40.40:5060;lr>

Record-Route: <sip:192.168.40.71:5060;lr>

Record-Route: <sip:rev.684527-192.168.40.71.dialog.cgatepro;lr>

From: <sip:7495-MANGO-NOMER@mangosip.ru>;tag=d4456a6d

To: <sip:SIP-ID-USER@sipnet.ru>;tag=as5f92b4f8

Call-ID: YmIyNjE2NTM5ODBjYzJmZjJlZDI4YjMzMzA5MTdkMzY.

CSeq: 2 INVITE

User-Agent: dlink 12-38-28928749-0.9.5.1.1140-IAD20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:SIP-ID-USER@EXT.IP.ASTERISK>

Content-Length: 0


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

[Kccall*CLI>

<--- SIP read from UDP://192.168.205.1:28874 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.205.33:5060;branch=z9hG4bK5220e964;rport=5060

Contact: <sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef>

To: <sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef>;tag=96462061

From: "7495-MANGO-NOMER"<sip:7495-MANGO-NOMER@192.168.205.33>;tag=as7d591f30

Call-ID: 522ff9010b926a495491d3be2dc6e8b6@192.168.205.33

CSeq: 102 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

Content-Type: application/sdp

User-Agent: eyeBeam release 1102u stamp 52345

Content-Length: 524

v=0

o=- 5 2 IN IP4 192.168.205.1

s=CounterPath eyeBeam 1.5

c=IN IP4 192.168.205.1

t=0 0

m=audio 34054 RTP/AVP 8 0 18 101

a=fmtp:18 annexb=yes

a=fmtp:101 0-15

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

m=video 32756 RTP/AVP 34 98 99

a=fmtp:34 QCIF=1 CIF=1 MaxBR=7180

a=fmtp:98 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=7180

a=fmtp:99 profile-level-id=42e01e; packetization-mode=1; max-br=718; max-mbps=11880

a=rtpmap:34 H263/90000

a=rtpmap:98 H263-1998/90000

a=rtpmap:99 H264/90000

a=sendrecv

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

[Kccall*CLI>
--- (11 headers 19 lines) ---

[Kccall*CLI>
Found RTP audio format 8
[Kccall*CLI>
Found RTP audio format 0
[Kccall*CLI>
Found RTP audio format 18
[Kccall*CLI>
Found RTP audio format 101
[Kccall*CLI>
Found RTP video format 34
[Kccall*CLI>
Found RTP video format 98
[Kccall*CLI>
Found RTP video format 99
[Kccall*CLI>
Peer audio RTP is at port 192.168.205.1:34054
[Kccall*CLI>
Found audio description format G729 for ID 18
[Kccall*CLI>
Found audio description format telephone-event for ID 101
[Kccall*CLI>
Found video description format H263 for ID 34
[Kccall*CLI>
Found video description format H263-1998 for ID 98
[Kccall*CLI>
Found video description format H264 for ID 99
[Kccall*CLI>
Capabilities: us - 0x38010f (g723|gsm|ulaw|alaw|g729|h263|h263p|h264), peer - audio=0x10c (ulaw|alaw|g729)/video=0x380000 (h263|h263p|h264)/text=0x0 (nothing), combined - 0x38010c (ulaw|alaw|g729|h263|h263p|h264)
[Kccall*CLI>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Kccall*CLI>
Peer audio RTP is at port 192.168.205.1:34054
[Kccall*CLI>
Peer video RTP is at port 192.168.205.1:32756
[Kccall*CLI>
list_route: hop: <sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef>
[Kccall*CLI>
set_destination: Parsing <sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef> for address/port to send to
[Kccall*CLI>
set_destination: set destination to 192.168.205.1, port 28874
[Kccall*CLI>
Transmitting (NAT) to 192.168.205.1:28874:
ACK sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef SIP/2.0

Via: SIP/2.0/UDP 192.168.205.33:5060;branch=z9hG4bK40c52e97;rport

Max-Forwards: 70

From: "7495-MANGO-NOMER" <sip:7495-MANGO-NOMER@192.168.205.33>;tag=as7d591f30

To: <sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef>;tag=96462061

Contact: <sip:7495-MANGO-NOMER@192.168.205.33>

Call-ID: 522ff9010b926a495491d3be2dc6e8b6@192.168.205.33

CSeq: 102 ACK

User-Agent: dlink 12-38-28928749-0.9.5.1.1140-IAD20

Content-Length: 0
---

[Kccall*CLI>
-- SIP/1001-08478ed8 answered SIP/SIP-ID-USER-083c0008
[Kccall*CLI>
Audio is at EXT.IP.ASTERISK port 11610
[Kccall*CLI>
Adding codec 0x8 (alaw) to SDP
[Kccall*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Kccall*CLI>

<--- Reliably Transmitting (NAT) to 212.53.40.40:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK833518-kmbdctj;cgp=etc.tario.ru;upaddr=81.88.80.36;received=212.53.40.40;rport=5060

Via: SIP/2.0/UDP 81.88.80.36;branch=z9hG4bKa6fe.e80a14020be51ff70c83500ef2acf327.0;i=4

Via: SIP/2.0/TCP 192.168.30.19:5060;branch=z9hG4bK-d8754z-60a0fc090951fa7a-1---d8754z-;rport=33068

Record-Route: <sip:212.53.40.40:5060;lr>

Record-Route: <sip:192.168.40.71:5060;lr>

Record-Route: <sip:rev.684527-192.168.40.71.dialog.cgatepro;lr>

From: <sip:7495-MANGO-NOMER@mangosip.ru>;tag=d4456a6d

To: <sip:SIP-ID-USER@sipnet.ru>;tag=as5f92b4f8

Call-ID: YmIyNjE2NTM5ODBjYzJmZjJlZDI4YjMzMzA5MTdkMzY.

CSeq: 2 INVITE

User-Agent: dlink 12-38-28928749-0.9.5.1.1140-IAD20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:SIP-ID-USER@EXT.IP.ASTERISK>

Content-Type: application/sdp

Content-Length: 254



v=0

o=root 55107865 55107865 IN IP4 EXT.IP.ASTERISK

s=Asterisk PBX 1.6.0.9

c=IN IP4 EXT.IP.ASTERISK

t=0 0

m=audio 11610 RTP/AVP 8 96

a=rtpmap:8 PCMA/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


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

---
[Kccall*CLI>
Retransmitting #6 (NAT) to 212.53.40.40:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK833518-kmbdctj;cgp=etc.tario.ru;upaddr=81.88.80.36;received=212.53.40.40;rport=5060

Via: SIP/2.0/UDP 81.88.80.36;branch=z9hG4bKa6fe.e80a14020be51ff70c83500ef2acf327.0;i=4

Via: SIP/2.0/TCP 192.168.30.19:5060;branch=z9hG4bK-d8754z-60a0fc090951fa7a-1---d8754z-;rport=33068

Record-Route: <sip:212.53.40.40:5060;lr>

Record-Route: <sip:192.168.40.71:5060;lr>

Record-Route: <sip:rev.684527-192.168.40.71.dialog.cgatepro;lr>

From: <sip:7495-MANGO-NOMER@mangosip.ru>;tag=d4456a6d

To: <sip:SIP-ID-USER@sipnet.ru>;tag=as5f92b4f8

Call-ID: YmIyNjE2NTM5ODBjYzJmZjJlZDI4YjMzMzA5MTdkMzY.

CSeq: 2 INVITE

User-Agent: dlink 12-38-28928749-0.9.5.1.1140-IAD20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:SIP-ID-USER@EXT.IP.ASTERISK>

Content-Type: application/sdp

Content-Length: 254



v=0

o=root 55107865 55107865 IN IP4 EXT.IP.ASTERISK

s=Asterisk PBX 1.6.0.9

c=IN IP4 EXT.IP.ASTERISK

t=0 0

m=audio 11610 RTP/AVP 8 96

a=rtpmap:8 PCMA/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


---

[Kccall*CLI>

<--- SIP read from UDP://192.168.205.1:28874 --->
<------------->

[Kccall*CLI>
[2010-03-09 22:27:15] [1;31;40mWARNING[0;37;40m[9696]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m2803[0;37;40m 1;37;40mretrans_pkt[0;37;40m: Maximum retries exceeded on transmission YmIyNjE2NTM5ODBjYzJmZjJlZDI4YjMzMzA5MTdkMzY. for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt.

[Kccall*CLI>
[2010-03-09 22:27:15] [1;31;40mWARNING[0;37;40m[9696]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m2830[0;37;40m [1;37;40mretrans_pkt[0;37;40m: Hanging up call YmIyNjE2NTM5ODBjYzJmZjJlZDI4YjMzMzA5MTdkMzY. - no reply to our critical packet (see doc/sip-retransmit.txt).

[Kccall*CLI>
Scheduling destruction of SIP dialog '522ff9010b926a495491d3be2dc6e8b6@192.168.205.33' in 32000 ms (Method: INVITE)
[Kccall*CLI>
set_destination: Parsing <sip:1001@192.168.205.1:28874;rinstance=b34ca42103b5f3ef> for address/port to send to
[Kccall*CLI>
set_destination: set destination to 192.168.205.1, port 28874
[Kccall*CLI>
Reliably Transmitting (NAT) to 192.168.205.1:28874:
BYE sip:1001@192.168.205.1:

Анализ мочи и кала. Урология, 2-й этаж, налево.

Вы намагнитили туда свой манго номер, и упорно всё сразу пытаетесь разрулить. Результат - ком проблем.
Нахрена звонить через неведомый для сипнета
From: <sip:7495-MANGO-NOMER@mangosip.ru> ??? sipnet.ru и отвечает - No user '7495-MANGO-NOMER' in SIP users list

Разбейте задачу на маленькие суб-задачи: например, дозвониться с одного SIP-ID-USER@sipnet.ru на тот, что зарегистрирован за вашим Астериском
To: <sip:SIP-ID-USER@sipnet.ru>

Если соединениние не будет рваться (а я подозреваю, что так), то проблема в стыке mangosip.ru и sipnet.ru. Не понимаю, зачем не отдавать этот номер из mangosip.ru в Астериск напрямую.

Да уж, действительно позвонил через номер доступа в Москве, на SIP-ID SIPNET и все было замечательно без обрывов.

Завтра поинтересуюсь у клиента, зачем ему была такая связка как MANGO -> SIPNET -> OFFICE.
Раньше у МАНГО только h323 был, но сейчас похоже они без проблем на SIP кидают.

Одно только смущает, почему все же X-Lite работает без проблем и без обрывов и можно ли что-то сделать с Астериском? А то как-то странно получается, если проблема Манго-Сип, то почему он не рвет связь и на X-Lite.

Спасибо, всем за помощь с проблемой, думаю решить ее именно настройкой MANGO -> OFFICE напрямую, без SIPNET.