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День добрый!!!
Может кто подскажет ответ на мой вопрос, в инете внятного ответа не нашел...
Ситуация такова:
Имеется GS-1004B подцепленый к SIP-серверу (3CX). Все работает замечательно, за исключением одного момента - при исходящем звонке через GSM (FXO не используется), если вызываемый абонент поднял трубку и после разговора положил трубку, то отбой проходит нормально ( SIP/2.0 486). Если же вызываемый абонент не поднимая трубки делает отбой вызова, то шлюз присылает SIP/2.0 480, SIP-сервер думает что канал не доступен, и начинает звонить по резервным маршрутам (в моем случае перебирает все 4 СИМ-карты и еще уходит на резервный городской номер).
Как его заставить передавать отбой в виде SIP/2.0 486?

Можете дебаги выложить одновременно в двух случая? (абонент берет трубку и просто отбивает звонок)
deb voip sip
deb voip call
deb rta ipc
deb gsm 0 0 call
deb gsm 0 0 rx
deb gsm 0 0 mon

вот debug voip sip
Вызывающий номер 7102
Вызываемый сотовый номер *7102
10.202.4.1 - SIP-сервер
10.202.4.3 - Addpac

1. Когда абонент отоветил и потом положил трубку. (нормально)

Received SIP PDU from ( 10.202.4.1:5060 )
INVITE sip:*7102@10.202.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-ef20fb70394c0b46-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10013@10.202.4.1:5060>
To: <sip:*7102@10.202.4.3:5060>
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719
Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 421

v=0
o=3cxPS 173140869120 249175212033 IN IP4 10.202.4.1
s=3cxPS Audio call
c=IN IP4 10.202.4.1
t=0 0
m=audio 7312 RTP/AVP 0 8 3 13 9 18 110 99 101
c=IN IP4 10.202.4.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 iLBC/8000
a=rtpmap:99 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv


Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-ef20fb70394c0b46-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719
To: <sip:*7102@10.202.4.3:5060>
Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk.
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0



Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-ef20fb70394c0b46-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719
To: <sip:*7102@10.202.4.3:5060>;tag=0e4f1e1fa4
Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk.
CSeq: 1 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:*7102@10.202.4.3
Content-Type: application/sdp
Content-Length: 240

v=0
o=*7102 1337265422 1337265422 IN IP4 10.202.4.3
s=AddPac Gateway SDP
c=IN IP4 10.202.4.3
t=1337265422 0
m=audio 23056 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv



Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-ef20fb70394c0b46-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719
To: <sip:*7102@10.202.4.3:5060>;tag=0e4f1e1fa4
Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk.
CSeq: 1 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:*7102@10.202.4.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 240

v=0
o=*7102 1337265432 1337265432 IN IP4 10.202.4.3
s=AddPac Gateway SDP
c=IN IP4 10.202.4.3
t=1337265432 0
m=audio 23056 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


Received SIP PDU from ( 10.202.4.1:5060 )
ACK sip:*7102@10.202.4.3 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-12218f51f9002765-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10013@10.202.4.1:5060>
To: <sip:*7102@10.202.4.3:5060>;tag=0e4f1e1fa4
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719
Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk.
CSeq: 1 ACK
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 0



Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
BYE sip:10013@10.202.4.1 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.3:5060;branch=z9hG4bK0e4f1e1fa45
From: <sip:*7102@10.202.4.3:5060>;tag=0e4f1e1fa4
To: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719
Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk.
CSeq: 5 BYE
Date: Thu, 17 May 2012 14:37:14 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:*7102@10.202.4.3>
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 10.202.4.1:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.4.3:5060;branch=z9hG4bK0e4f1e1fa45
Contact: <sip:10013@10.202.4.1:5060>
To: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719
From: <sip:*7102@10.202.4.3:5060>;tag=0e4f1e1fa4
Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk.
CSeq: 5 BYE
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 0





2. Когда вызываемый абонент сразу отбил бызов. (не нормально)


Received SIP PDU from ( 10.202.4.1:5060 )
INVITE sip:*7102@10.202.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-481430690248db03-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10013@10.202.4.1:5060>
To: <sip:*7102@10.202.4.3:5060>
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=983b1925
Call-ID: MjRhYThkZGZmNGRkY2Q2ODQyZjNiNzM2OTYyYjAzYTk.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 420

v=0
o=3cxPS 12448694272 290614935553 IN IP4 10.202.4.1
s=3cxPS Audio call
c=IN IP4 10.202.4.1
t=0 0
m=audio 7278 RTP/AVP 0 8 3 13 9 18 110 99 101
c=IN IP4 10.202.4.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 iLBC/8000
a=rtpmap:99 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv


Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-481430690248db03-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=983b1925
To: <sip:*7102@10.202.4.3:5060>
Call-ID: MjRhYThkZGZmNGRkY2Q2ODQyZjNiNzM2OTYyYjAzYTk.
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0



Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-481430690248db03-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=983b1925
To: <sip:*7102@10.202.4.3:5060>;tag=2f4f3b1ba4
Call-ID: MjRhYThkZGZmNGRkY2Q2ODQyZjNiNzM2OTYyYjAzYTk.
CSeq: 1 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:*7102@10.202.4.3
Content-Type: application/sdp
Content-Length: 240

v=0
o=*7102 1337265199 1337265199 IN IP4 10.202.4.3
s=AddPac Gateway SDP
c=IN IP4 10.202.4.3
t=1337265199 0
m=audio 23048 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-481430690248db03-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=983b1925
To: <sip:*7102@10.202.4.3:5060>;tag=2f4f3b1ba4
Call-ID: MjRhYThkZGZmNGRkY2Q2ODQyZjNiNzM2OTYyYjAzYTk.
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0



Received SIP PDU from ( 10.202.4.1:5060 )
ACK sip:*7102@10.202.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-481430690248db03-1---d8754z-;rport
Max-Forwards: 70
To: <sip:*7102@10.202.4.3:5060>;tag=2f4f3b1ba4
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=983b1925
Call-ID: MjRhYThkZGZmNGRkY2Q2ODQyZjNiNzM2OTYyYjAzYTk.
CSeq: 1 ACK
Content-Length: 0



Received SIP PDU from ( 10.202.4.1:5060 )
INVITE sip:*7102@10.202.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-d362384179604b46-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10013@10.202.4.1:5060>
To: <sip:*7102@10.202.4.3:5060>
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=7c20da3e
Call-ID: MGM5MjliMGU4Mzc4MzIyZGUxYzBhYTA2NGM5NGZlOTQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 419

v=0
o=3cxPS 3657433088 297191604225 IN IP4 10.202.4.1
s=3cxPS Audio call
c=IN IP4 10.202.4.1
t=0 0
m=audio 7280 RTP/AVP 0 8 3 13 9 18 110 99 101
c=IN IP4 10.202.4.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 iLBC/8000
a=rtpmap:99 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv


Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-d362384179604b46-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=7c20da3e
To: <sip:*7102@10.202.4.3:5060>
Call-ID: MGM5MjliMGU4Mzc4MzIyZGUxYzBhYTA2NGM5NGZlOTQ.
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0



Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-d362384179604b46-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=7c20da3e
To: <sip:*7102@10.202.4.3:5060>;tag=394fea1ca4
Call-ID: MGM5MjliMGU4Mzc4MzIyZGUxYzBhYTA2NGM5NGZlOTQ.
CSeq: 1 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:*7102@10.202.4.3
Content-Type: application/sdp
Content-Length: 240

v=0
o=*7102 1337265209 1337265209 IN IP4 10.202.4.3
s=AddPac Gateway SDP
c=IN IP4 10.202.4.3
t=1337265209 0
m=audio 23050 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv








чуть позже другие дебаги сделаю...

вот debug voip call

Выриант 1 (нормальный)

1 <Call 49> : ****** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337266676) ****
2 <SIP 49> : Receive INVITE Request
3 <NetCon 49> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1)
4 <NetCon 49> : Found inbound voip peer by dest-pattern id(12201)
5 <NetCon 49> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3)
6 <Call 49> : From Net - calledParty(*7102) callingParty(10013)
7 <Call 49> : MatchedPerfect
8 <Call 49> : MatchAllProcess After Sorted
<0> id(4589) dest(*....) prefer(0) selected(5)
<1> id(4590) dest(*....) prefer(0) selected(5)
<2> id(4591) dest(*....) prefer(0) selected(5)
<3> id(4588) dest(*....) prefer(0) selected(6)
9 <Call 49> : Initiate callee with dial-peer(*....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
10 <CEP 010100> : InitiateOutCall : calledNum(*7102), callingNum(10013), callerPort(ffffffff) type(GSM)
11 <CEP 010100> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(49)
12 <SIP 49> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
13 <SIP 49> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
14 <PhonePlay 49> : Audio Count(1)
15 <PhonePlay 49> : rtpSessionId(1) Second Audio Port(-1)
16 <SIP 49> : SetAlerting
17 <Call 49> : PreConnected from(10100)
18 <SIP 49> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
19 <SIP 49> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
20 <SIP 49> : Add Local Audio MediaFormat : 0
21 <Call 49> : Connected from(10100)
22 <SIP 49> : SetConnected
23 <SIP 49> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
24 <SIP 49> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
25 <SIP 49> : Add Local Audio MediaFormat : 0
26 <SIP 49> : ACK received
27 <SIP 49> : Receive ACK Request
28 <SIP 49> : Set Terminated Success for 1 INVITE
29 <CEP 010100> : Disconnected(16) at Busy
30 <Call 49> : Terminated from(10100) this(Local:CallClear) before((null)) forced(0) time(1337266688)
31 <SIP 49> : ReleaseWithBYE
32 <SIP 49> : Send BYE Request
33 <NetEP 49> : Call FROM <Sergey> terminated reason(Local:CallClear)
34 <CEP 010100> : DisconnectCall at Idle
35 <SIP 49> : Receive 200 OK
36 <SIP 49> : Transaction (6 BYE) completed



Выриант 2(не нормальный)

1 <Call 47> : ****** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337266569) ****
2 <SIP 47> : Receive INVITE Request
3 <NetCon 47> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1)
4 <NetCon 47> : Found inbound voip peer by dest-pattern id(12201)
5 <NetCon 47> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3)
6 <Call 47> : From Net - calledParty(*7102) callingParty(10013)
7 <Call 47> : MatchedPerfect
8 <Call 47> : MatchAllProcess After Sorted
<0> id(4591) dest(*....) prefer(0) selected(4)
<1> id(4588) dest(*....) prefer(0) selected(5)
<2> id(4589) dest(*....) prefer(0) selected(5)
<3> id(4590) dest(*....) prefer(0) selected(5)
9 <Call 47> : Initiate callee with dial-peer(*....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
10 <CEP 010300> : InitiateOutCall : calledNum(*7102), callingNum(10013), callerPort(ffffffff) type(GSM)
11 <CEP 010300> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(47)
12 <SIP 47> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
13 <SIP 47> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
14 <PhonePlay 47> : Audio Count(1)
15 <PhonePlay 47> : rtpSessionId(1) Second Audio Port(-1)
16 <SIP 47> : SetAlerting
17 <Call 47> : PreConnected from(10300)
18 <SIP 47> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
19 <SIP 47> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
20 <SIP 47> : Add Local Audio MediaFormat : 0
21 <CEP 010300> : Disconnected(16) at Busy
22 <Call 47> : Terminated from(10300) this(Local:CallClear) before((null)) forced(0) time(1337266578)
23 <NetEP 47> : Call FROM <Sergey> terminated reason(Local:CallClear)
24 <CEP 010300> : DisconnectCall at Idle
25 <SIP 47> : Receive ACK Request
26 <Call 48> : ****** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337266578) ****
27 <SIP 48> : Receive INVITE Request
28 <NetCon 48> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1)
29 <NetCon 48> : Found inbound voip peer by dest-pattern id(12201)
30 <NetCon 48> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3)
31 <Call 48> : From Net - calledParty(*7102) callingParty(10013)
32 <Call 48> : MatchedPerfect
33 <Call 48> : MatchAllProcess After Sorted
<0> id(4588) dest(*....) prefer(0) selected(5)
<1> id(4589) dest(*....) prefer(0) selected(5)
<2> id(4590) dest(*....) prefer(0) selected(5)
<3> id(4591) dest(*....) prefer(0) selected(5)
34 <Call 48> : Initiate callee with dial-peer(*....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
35 <CEP 010000> : InitiateOutCall : calledNum(*7102), callingNum(10013), callerPort(ffffffff) type(GSM)
36 <CEP 010000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(48)
37 <SIP 48> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
38 <SIP 48> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
39 <PhonePlay 48> : Audio Count(1)
40 <PhonePlay 48> : rtpSessionId(1) Second Audio Port(-1)
41 <SIP 48> : SetAlerting
42 <Call 48> : PreConnected from(10000)
43 <SIP 48> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
44 <SIP 48> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
45 <SIP 48> : Add Local Audio MediaFormat : 0

Вот debug rta ipc

Вариант 1 (нормальный)

[2891.255] RTA(1/0/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1)
[2891.255] VP(1/0/0) open channel
[2891.255] VM(1/0/0) Tx GSM CallRequest 1 stage *7102
[2891.255] VM(1/0/0) set T38 enable by CCC
[2891.255] VM(1/0/0) set T38 mode STD
[2891.255] VM(1/0/0) Fax rate 9600
[2891.255] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[2891.255] VM(1/0/0) VAD disable
[2891.255] VP(1/0/0) update VAD 0
[2891.255] VM(1/0/0) SID enable by CCC
[2891.270] RTA(1/0/0) Rx RS_OPEN_REQ callId=52 ssId=1 G711U
peer=10.202.4.1 mp=23104/23105 hp=7050/7051
[2891.270] VM(1/0/0) codec same G711U
[2891.270] RTA(1/0/0) Rx RS_LISTEN_REQ callId=52 ssId=1 G711U
peer=10.202.4.1 mp=23104/23105 hp=7050/7051
[2899.055] RTA(1/0/0) Rx GSM_STTS_IND CALL_CONN
[2899.055] RTA(1/0/0) Rx GSM_STTS_CALL_CONN at state=5
[2899.055] VP(1/0/0) attribute Fax enable, Modem disable
[2899.055] VP(1/0/0) update Fax enable, Modem disable
[2899.055] VM(1/0/0) Tx CONNECT_CNF
[2899.055] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[2899.055] VM(1/0/0) VAD disable
[2899.055] VM(1/0/0) SID enable by CCC
[2899.070] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[2899.070] VM(1/0/0) DTMF_RTP_RFC2833 enable
[2899.070] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65
[2901.780] RTA(1/0/0) Rx GSM_STTS_IND CALL_DISC
[2901.780] RTA(1/0/0) Rx GSM_STTS_CALL_DISC at state=6
[2901.780] VM(1/0/0) vopp idle
[2901.780] VP(1/0/0) close channel
[2901.780] VM(1/0/0) Tx DISCONN_CNF
[2901.790] RTA(1/0/0) Rx RS_CLOSE_REQ callId=52 ssId=1 dir=all
[2901.790] RTA(1/0/0) close Media socket
[2901.790] RTA(1/0/0) close RTCP socket



Вариант 2 (не нормальный)

[2810.065] RTA(1/2/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1)
[2810.065] VP(1/2/0) open channel
[2810.065] VM(1/2/0) Tx GSM CallRequest 1 stage *7102
[2810.065] VM(1/2/0) set T38 enable by CCC
[2810.065] VM(1/2/0) set T38 mode STD
[2810.065] VM(1/2/0) Fax rate 9600
[2810.065] RTA(1/2/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[2810.065] VM(1/2/0) VAD disable
[2810.065] VP(1/2/0) update VAD 0
[2810.065] VM(1/2/0) SID enable by CCC
[2810.080] RTA(1/2/0) Rx RS_OPEN_REQ callId=50 ssId=1 G711U
peer=10.202.4.1 mp=23100/23101 hp=7040/7041
[2810.080] VM(1/2/0) codec same G711U
[2810.080] RTA(1/2/0) Rx RS_LISTEN_REQ callId=50 ssId=1 G711U
peer=10.202.4.1 mp=23100/23101 hp=7040/7041
[2818.665] RTA(1/2/0) Rx GSM_STTS_IND CALL_DISC
[2818.665] RTA(1/2/0) Rx GSM_STTS_CALL_DISC at state=5
[2818.665] VM(1/2/0) vopp idle
[2818.665] VP(1/2/0) close channel
[2818.665] VM(1/2/0) Tx DISCONN_CNF
[2818.665] RTA(1/2/0) Rx RS_CLOSE_REQ callId=50 ssId=1 dir=all
[2818.665] RTA(1/2/0) close Media socket
[2818.665] RTA(1/2/0) close RTCP socket
[2818.795] RTA(1/3/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1)
[2818.795] VP(1/3/0) open channel
[2818.795] VM(1/3/0) Tx GSM CallRequest 1 stage *7102
[2818.795] VM(1/3/0) set T38 enable by CCC
[2818.795] VM(1/3/0) set T38 mode STD
[2818.795] VM(1/3/0) Fax rate 9600
[2818.800] RTA(1/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[2818.800] VM(1/3/0) VAD disable
[2818.800] VP(1/3/0) update VAD 0
[2818.800] VM(1/3/0) SID enable by CCC
[2818.810] RTA(1/3/0) Rx RS_OPEN_REQ callId=51 ssId=1 G711U
peer=10.202.4.1 mp=23102/23103 hp=7042/7043
[2818.810] VM(1/3/0) codec same G711U
[2818.810] RTA(1/3/0) Rx RS_LISTEN_REQ callId=51 ssId=1 G711U
peer=10.202.4.1 mp=23102/23103 hp=7042/7043

deb gsm 1 0 rx

Вариант 1 (нормальный)

GSM-1/0-RSP: [0,0] 0D
GSM-1/0-RSP: [0,0] 0A
GSM-1/0-RSP: [0,0] 2B
GSM-1/0-RSP: [0,1] 57
GSM-1/0-RSP: [0,2] 49
GSM-1/0-RSP: [0,3] 4E
GSM-1/0-RSP: [0,4] 44
GSM-1/0-RSP: [0,5] 3A
GSM-1/0-RSP: [0,6] 20
GSM-1/0-RSP: [0,7] 35
GSM-1/0-RSP: [0,8] 2C
GSM-1/0-RSP: [0,9] 31
GSM-1/0-RSP: [0,10] 0D
GSM-1/0-RSP: [0,10] 0A
[3325.826] GSM-1/0: RECV, Line 0 : +WIND: 5,1
[3325.826] GSM-1/0: RECV, Line 0 : 2B 57 49 4E 44 3A 20 35 2C 31
GSM-1/0-RSP: [1,0] 0D
GSM-1/0-RSP: [1,0] 0A
GSM-1/0-RSP: [1,0] 2B
GSM-1/0-RSP: [1,1] 57
GSM-1/0-RSP: [1,2] 49
GSM-1/0-RSP: [1,3] 4E
GSM-1/0-RSP: [1,4] 44
GSM-1/0-RSP: [1,5] 3A
GSM-1/0-RSP: [1,6] 20
GSM-1/0-RSP: [1,7] 32
GSM-1/0-RSP: [1,8] 0D
GSM-1/0-RSP: [1,8] 0A
[3332.971] GSM-1/0: RECV, Line 1 : +WIND: 2
[3332.971] GSM-1/0: RECV, Line 1 : 2B 57 49 4E 44 3A 20 32
GSM-1/0-RSP: [2,0] 0D
GSM-1/0-RSP: [2,0] 0A
GSM-1/0-RSP: [2,0] 4F
GSM-1/0-RSP: [2,1] 4B
GSM-1/0-RSP: [2,2] 0D
GSM-1/0-RSP: [2,2] 0A
[3334.380] GSM-1/0: RECV, Line 2 : OK
[3334.380] GSM-1/0: RECV, Line 2 : 4F 4B
GSM-1/0-EVT: [0] 0D
GSM-1/0-EVT: [0] 0A
GSM-1/0-EVT: [0] 4E
GSM-1/0-EVT: [1] 4F
GSM-1/0-EVT: [2] 20
GSM-1/0-EVT: [3] 43
GSM-1/0-EVT: [4] 41
GSM-1/0-EVT: [5] 52
GSM-1/0-EVT: [6] 52
GSM-1/0-EVT: [7] 49
GSM-1/0-EVT: [8] 45
GSM-1/0-EVT: [9] 52
GSM-1/0-EVT: [10] 0D
GSM-1/0-EVT: [10] 0A
GSM-1/0-EVT: [0] 0D
GSM-1/0-EVT: [0] 0A
GSM-1/0-EVT: [0] 2B
GSM-1/0-EVT: [1] 57
GSM-1/0-EVT: [2] 49
GSM-1/0-EVT: [3] 4E
GSM-1/0-EVT: [4] 44
GSM-1/0-EVT: [5] 3A
GSM-1/0-EVT: [6] 20
GSM-1/0-EVT: [7] 36
GSM-1/0-EVT: [8] 2C
GSM-1/0-EVT: [9] 31
GSM-1/0-EVT: [10] 0D
GSM-1/0-EVT: [10] 0A
GSM-1/0-RSP: [0,0] 0D
GSM-1/0-RSP: [0,0] 0A
GSM-1/0-RSP: [0,0] 2B
GSM-1/0-RSP: [0,1] 43
GSM-1/0-RSP: [0,2] 53
GSM-1/0-RSP: [0,3] 51
GSM-1/0-RSP: [0,4] 3A
GSM-1/0-RSP: [0,5] 20
GSM-1/0-RSP: [0,6] 31
GSM-1/0-RSP: [0,7] 35
GSM-1/0-RSP: [0,8] 2C
GSM-1/0-RSP: [0,9] 30
GSM-1/0-RSP: [0,10] 0D
GSM-1/0-RSP: [0,10] 0A
[3339.772] GSM-1/0: RECV, Line 0 : +CSQ: 15,0
[3339.772] GSM-1/0: RECV, Line 0 : 2B 43 53 51 3A 20 31 35 2C 30
GSM-1/0-RSP: [1,0] 0D
GSM-1/0-RSP: [1,0] 0A
GSM-1/0-RSP: [1,0] 4F
GSM-1/0-RSP: [1,1] 4B
GSM-1/0-RSP: [1,2] 0D
GSM-1/0-RSP: [1,2] 0A
[3339.772] GSM-1/0: RECV, Line 1 : OK
[3339.772] GSM-1/0: RECV, Line 1 : 4F 4B



Вариант 2 (не нормальный)

GSM-1/2-RSP: [0,0] 0D
GSM-1/2-RSP: [0,0] 0A
GSM-1/2-RSP: [0,0] 2B
GSM-1/2-RSP: [0,1] 57
GSM-1/2-RSP: [0,2] 49
GSM-1/2-RSP: [0,3] 4E
GSM-1/2-RSP: [0,4] 44
GSM-1/2-RSP: [0,5] 3A
GSM-1/2-RSP: [0,6] 20
GSM-1/2-RSP: [0,7] 35
GSM-1/2-RSP: [0,8] 2C
GSM-1/2-RSP: [0,9] 31
GSM-1/2-RSP: [0,10] 0D
GSM-1/2-RSP: [0,10] 0A
[3227.368] GSM-1/2: RECV, Line 0 : +WIND: 5,1
[3227.368] GSM-1/2: RECV, Line 0 : 2B 57 49 4E 44 3A 20 35 2C 31
GSM-1/2-RSP: [1,0] 0D
GSM-1/2-RSP: [1,0] 0A
GSM-1/2-RSP: [1,0] 2B
GSM-1/2-RSP: [1,1] 57
GSM-1/2-RSP: [1,2] 49
GSM-1/2-RSP: [1,3] 4E
GSM-1/2-RSP: [1,4] 44
GSM-1/2-RSP: [1,5] 3A
GSM-1/2-RSP: [1,6] 20
GSM-1/2-RSP: [1,7] 32
GSM-1/2-RSP: [1,8] 0D
GSM-1/2-RSP: [1,8] 0A
[3234.627] GSM-1/2: RECV, Line 1 : +WIND: 2
[3234.627] GSM-1/2: RECV, Line 1 : 2B 57 49 4E 44 3A 20 32
GSM-1/2-RSP: [2,0] 0D
GSM-1/2-RSP: [2,0] 0A
GSM-1/2-RSP: [2,0] 42
GSM-1/2-RSP: [2,1] 55
GSM-1/2-RSP: [2,2] 53
GSM-1/2-RSP: [2,3] 59
GSM-1/2-RSP: [2,4] 0D
GSM-1/2-RSP: [2,4] 0A
[3238.043] GSM-1/2: RECV, Line 2 : BUSY
[3238.043] GSM-1/2: RECV, Line 2 : 42 55 53 59
GSM-1/2-RSP: [3,0] 0D
GSM-1/2-RSP: [3,0] 0A
GSM-1/2-RSP: [3,0] 2B
GSM-1/2-RSP: [3,1] 57
GSM-1/2-RSP: [3,2] 49
GSM-1/2-RSP: [3,3] 4E
GSM-1/2-RSP: [3,4] 44
GSM-1/2-RSP: [3,5] 3A
GSM-1/2-RSP: [3,6] 20
GSM-1/2-RSP: [3,7] 36
GSM-1/2-RSP: [3,8] 2C
GSM-1/2-RSP: [3,9] 31
GSM-1/2-RSP: [3,10] 0D
GSM-1/2-RSP: [3,10] 0A
[3238.046] GSM-1/2: RECV, Line 3 : +WIND: 6,1
[3238.046] GSM-1/2: RECV, Line 3 : 2B 57 49 4E 44 3A 20 36 2C 31
GSM-1/2-RSP: [0,0] 0D
GSM-1/2-RSP: [0,0] 0A
GSM-1/2-RSP: [0,0] 4F
GSM-1/2-RSP: [0,1] 4B
GSM-1/2-RSP: [0,2] 0D
GSM-1/2-RSP: [0,2] 0A
[3238.142] GSM-1/2: RECV, Line 0 : OK
[3238.142] GSM-1/2: RECV, Line 0 : 4F 4B

GSM-1/2-RSP: [0,0] 0D
GSM-1/2-RSP: [0,0] 0A
GSM-1/2-RSP: [0,0] 2B
GSM-1/2-RSP: [0,1] 43
GSM-1/2-RSP: [0,2] 53
GSM-1/2-RSP: [0,3] 51
GSM-1/2-RSP: [0,4] 3A
GSM-1/2-RSP: [0,5] 20
GSM-1/2-RSP: [0,6] 32
GSM-1/2-RSP: [0,7] 31
GSM-1/2-RSP: [0,8] 2C
GSM-1/2-RSP: [0,9] 34
GSM-1/2-RSP: [0,10] 0D
GSM-1/2-RSP: [0,10] 0A
[3244.773] GSM-1/2: RECV, Line 0 : +CSQ: 21,4
[3244.773] GSM-1/2: RECV, Line 0 : 2B 43 53 51 3A 20 32 31 2C 34
GSM-1/2-RSP: [1,0] 0D
GSM-1/2-RSP: [1,0] 0A
GSM-1/2-RSP: [1,0] 4F
GSM-1/2-RSP: [1,1] 4B
GSM-1/2-RSP: [1,2] 0D
GSM-1/2-RSP: [1,2] 0A
[3244.773] GSM-1/2: RECV, Line 1 : OK
[3244.773] GSM-1/2: RECV, Line 1 : 4F 4B

Дебаги нужны одновременно.
И по одному порту, чтоб лишние звонки не мешали.

Ок

Вариант 1 (нормальный)

Received SIP PDU from ( 10.202.4.1:5060 )
INVITE sip:*7102@10.202.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-e108521059601175-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10013@10.202.4.1:5060>
To: <sip:*7102@10.202.4.3:5060>
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32
Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 421

v=0
o=3cxPS 378661765120 211023822849 IN IP4 10.202.4.1
s=3cxPS Audio call
c=IN IP4 10.202.4.1
t=0 0
m=audio 7094 RTP/AVP 0 8 3 13 9 18 110 99 101
c=IN IP4 10.202.4.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 iLBC/8000
a=rtpmap:99 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv


Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-e108521059601175-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32
To: <sip:*7102@10.202.4.3:5060>
Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU.
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


1 <Call 585> : ****** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337874686) ****
2 <SIP 585> : Receive INVITE Request
3 <NetCon 585> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1)
4 <NetCon 585> : Found inbound voip peer by dest-pattern id(12201)
5 <NetCon 585> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3)
6 <Call 585> : From Net - calledParty(*7102) callingParty(10013)
7 <Call 585> : MatchedPerfect
8 <Call 585> : MatchAllProcess After Sorted
<0> id(4589) dest(*....) prefer(0) selected(7)
<1> id(4590) dest(*....) prefer(0) selected(7)
<2> id(4591) dest(*....) prefer(0) selected(7)
<3> id(4588) dest(*....) prefer(0) selected(8)
9 <Call 585> : Initiate callee with dial-peer(*....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
10 <CEP 010100> : InitiateOutCall : calledNum(*7102), callingNum(10013), callerPort(ffffffff) type(GSM)
[165500.400] RTA(1/1/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1)
[165500.405] VP(1/1/0) open channel
[165500.405] VM(1/1/0) Tx GSM CallRequest 1 stage *7102
11 <CEP 010100> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(585)
[165500.405] VM(1/1/0) set T38 enable by CCC
[165500.405] VM(1/1/0) set T38 mode STD
[165500.405] VM(1/1/0) Fax rate 9600
12 <SIP 585> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
13 <SIP 585> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
14 <PhonePlay 585> : Audio Count(1)
15 <PhonePlay 585> : rtpSessionId(1) Second Audio Port(-1)
16 <SIP 585> : SetAlerting
17 <Call 585> : PreConnected from(10100)
[165500.405] RTA(1/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[165500.405] VM(1/1/0) VAD disable
[165500.405] VP(1/1/0) update VAD 0
[165500.405] VM(1/1/0) SID enable by CCC
18 <SIP 585> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
19 <SIP 585> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
20 <SIP 585> : Add Local Audio MediaFormat : 0

Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-e108521059601175-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32
To: <sip:*7102@10.202.4.3:5060>;tag=fe4f374ea4
Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU.
CSeq: 1 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:*7102@10.202.4.3
Content-Type: application/sdp
Content-Length: 240

v=0
o=*7102 1337874686 1337874686 IN IP4 10.202.4.3
s=AddPac Gateway SDP
c=IN IP4 10.202.4.3
t=1337874686 0
m=audio 24170 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

[165500.415] RTA(1/1/0) Rx RS_OPEN_REQ callId=585 ssId=1 G711U
peer=10.202.4.1 mp=24170/24171 hp=7094/7095
[165500.420] VM(1/1/0) codec same G711U
[165500.420] RTA(1/1/0) Rx RS_LISTEN_REQ callId=585 ssId=1 G711U
peer=10.202.4.1 mp=24170/24171 hp=7094/7095
[165509.205] RTA(1/1/0) Rx GSM_STTS_IND CALL_CONN
[165509.205] RTA(1/1/0) Rx GSM_STTS_CALL_CONN at state=5
[165509.205] VP(1/1/0) attribute Fax enable, Modem disable
[165509.205] VP(1/1/0) update Fax enable, Modem disable
[165509.205] VM(1/1/0) Tx CONNECT_CNF
21 <Call 585> : Connected from(10100)
[165509.205] RTA(1/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[165509.205] VM(1/1/0) VAD disable
[165509.205] VM(1/1/0) SID enable by CCC
22 <SIP 585> : SetConnected
23 <SIP 585> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
24 <SIP 585> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
25 <SIP 585> : Add Local Audio MediaFormat : 0

Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-e108521059601175-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32
To: <sip:*7102@10.202.4.3:5060>;tag=fe4f374ea4
Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU.
CSeq: 1 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:*7102@10.202.4.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 240

v=0
o=*7102 1337874695 1337874695 IN IP4 10.202.4.3
s=AddPac Gateway SDP
c=IN IP4 10.202.4.3
t=1337874695 0
m=audio 24170 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

[165509.220] RTA(1/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[165509.220] VM(1/1/0) DTMF_RTP_RFC2833 enable
[165509.220] RTA(1/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65

Received SIP PDU from ( 10.202.4.1:5060 )
ACK sip:*7102@10.202.4.3 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-804aad5b2738457d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10013@10.202.4.1:5060>
To: <sip:*7102@10.202.4.3:5060>;tag=fe4f374ea4
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32
Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU.
CSeq: 1 ACK
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 0


26 <SIP 585> : ACK received
27 <SIP 585> : Receive ACK Request
28 <SIP 585> : Set Terminated Success for 1 INVITE
[165512.545] RTA(1/1/0) Rx GSM_STTS_IND CALL_DISC
[165512.545] RTA(1/1/0) Rx GSM_STTS_CALL_DISC at state=6
[165512.545] VM(1/1/0) vopp idle
[165512.545] VP(1/1/0) close channel
[165512.545] VM(1/1/0) Tx DISCONN_CNF
29 <CEP 010100> : Disconnected(16) at Busy
30 <Call 585> : Terminated from(10100) this(Local:CallClear) before((null)) forced(0) time(1337874698)
31 <SIP 585> : ReleaseWithBYE
32 <SIP 585> : Send BYE Request

Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
BYE sip:10013@10.202.4.1 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.3:5060;branch=z9hG4bKfe4f374ea497
From: <sip:*7102@10.202.4.3:5060>;tag=fe4f374ea4
To: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32
Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU.
CSeq: 97 BYE
Date: Thu, 24 May 2012 15:51:38 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:*7102@10.202.4.3>
Content-Length: 0
Max-Forwards: 70


[165512.555] RTA(1/1/0) Rx RS_CLOSE_REQ callId=585 ssId=1 dir=all
[165512.555] RTA(1/1/0) close Media socket
[165512.555] RTA(1/1/0) close RTCP socket
33 <NetEP 585> : Call FROM <Sergey> terminated reason(Local:CallClear)
34 <CEP 010100> : DisconnectCall at Idle

Received SIP PDU from ( 10.202.4.1:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.4.3:5060;branch=z9hG4bKfe4f374ea497
Contact: <sip:10013@10.202.4.1:5060>
To: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32
From: <sip:*7102@10.202.4.3:5060>;tag=fe4f374ea4
Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU.
CSeq: 97 BYE
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 0


35 <SIP 585> : Receive 200 OK
36 <SIP 585> : Transaction (97 BYE) completed



Вариант 2 (не нормальный)

Received SIP PDU from ( 10.202.4.1:5060 )
INVITE sip:*7102@10.202.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-4678e870be074e64-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10013@10.202.4.1:5060>
To: <sip:*7102@10.202.4.3:5060>
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=74195a00
Call-ID: MTc4NjQ0NmVmNTlkMTg5MDY1YTg3ZWRjZDZjODJmYjE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 419

v=0
o=3cxPS 81117839360 21625831425 IN IP4 10.202.4.1
s=3cxPS Audio call
c=IN IP4 10.202.4.1
t=0 0
m=audio 7158 RTP/AVP 0 8 3 13 9 18 110 99 101
c=IN IP4 10.202.4.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 iLBC/8000
a=rtpmap:99 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv


Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-4678e870be074e64-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=74195a00
To: <sip:*7102@10.202.4.3:5060>
Call-ID: MTc4NjQ0NmVmNTlkMTg5MDY1YTg3ZWRjZDZjODJmYjE.
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


2 <Call 591> : ****** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337875310) ****
3 <SIP 591> : Receive INVITE Request
4 <NetCon 591> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1)
5 <NetCon 591> : Found inbound voip peer by dest-pattern id(12201)
6 <NetCon 591> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3)
7 <Call 591> : From Net - calledParty(*7102) callingParty(10013)
8 <Call 591> : MatchedPerfect
9 <Call 591> : MatchAllProcess After Sorted
<0> id(4591) dest(*....) prefer(0) selected(8)
<1> id(4588) dest(*....) prefer(0) selected(9)
<2> id(4589) dest(*....) prefer(0) selected(9)
<3> id(4590) dest(*....) prefer(0) selected(9)
10 <Call 591> : Initiate callee with dial-peer(*....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
11 <CEP 010300> : InitiateOutCall : calledNum(*7102), callingNum(10013), callerPort(ffffffff) type(GSM)
[166127.515] RTA(1/3/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1)
[166127.515] VP(1/3/0) open channel
[166127.515] VM(1/3/0) Tx GSM CallRequest 1 stage *7102
12 <CEP 010300> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(591)
[166127.515] VM(1/3/0) set T38 enable by CCC
[166127.515] VM(1/3/0) set T38 mode STD
[166127.520] VM(1/3/0) Fax rate 9600
13 <SIP 591> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
14 <SIP 591> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
15 <PhonePlay 591> : Audio Count(1)
16 <PhonePlay 591> : rtpSessionId(1) Second Audio Port(-1)
17 <SIP 591> : SetAlerting
18 <Call 591> : PreConnected from(10300)
[166127.520] RTA(1/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[166127.520] VM(1/3/0) VAD disable
[166127.520] VP(1/3/0) update VAD 0
[166127.520] VM(1/3/0) SID enable by CCC
19 <SIP 591> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
20 <SIP 591> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
21 <SIP 591> : Add Local Audio MediaFormat : 0

Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-4678e870be074e64-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=74195a00
To: <sip:*7102@10.202.4.3:5060>;tag=6e4f3554a4
Call-ID: MTc4NjQ0NmVmNTlkMTg5MDY1YTg3ZWRjZDZjODJmYjE.
CSeq: 1 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:*7102@10.202.4.3
Content-Type: application/sdp
Content-Length: 240

v=0
o=*7102 1337875310 1337875310 IN IP4 10.202.4.3
s=AddPac Gateway SDP
c=IN IP4 10.202.4.3
t=1337875310 0
m=audio 24182 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

[166127.530] RTA(1/3/0) Rx RS_OPEN_REQ callId=591 ssId=1 G711U
peer=10.202.4.1 mp=24182/24183 hp=7158/7159
[166127.530] VM(1/3/0) codec same G711U
[166127.530] RTA(1/3/0) Rx RS_LISTEN_REQ callId=591 ssId=1 G711U
peer=10.202.4.1 mp=24182/24183 hp=7158/7159
[166136.115] RTA(1/3/0) Rx GSM_STTS_IND CALL_DISC
[166136.115] RTA(1/3/0) Rx GSM_STTS_CALL_DISC at state=5
[166136.115] VM(1/3/0) vopp idle
[166136.115] VP(1/3/0) close channel
[166136.115] VM(1/3/0) Tx DISCONN_CNF
22 <CEP 010300> : Disconnected(16) at Busy
23 <Call 591> : Terminated from(10300) this(Local:CallClear) before((null)) forced(0) time(1337875319)

Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-4678e870be074e64-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=74195a00
To: <sip:*7102@10.202.4.3:5060>;tag=6e4f3554a4
Call-ID: MTc4NjQ0NmVmNTlkMTg5MDY1YTg3ZWRjZDZjODJmYjE.
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


[166136.115] RTA(1/3/0) Rx RS_CLOSE_REQ callId=591 ssId=1 dir=all
[166136.115] RTA(1/3/0) close Media socket
[166136.115] RTA(1/3/0) close RTCP socket
24 <NetEP 591> : Call FROM <Sergey> terminated reason(Local:CallClear)
25 <CEP 010300> : DisconnectCall at Idle

Received SIP PDU from ( 10.202.4.1:5060 )
ACK sip:*7102@10.202.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-4678e870be074e64-1---d8754z-;rport
Max-Forwards: 70
To: <sip:*7102@10.202.4.3:5060>;tag=6e4f3554a4
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=74195a00
Call-ID: MTc4NjQ0NmVmNTlkMTg5MDY1YTg3ZWRjZDZjODJmYjE.
CSeq: 1 ACK
Content-Length: 0


26 <SIP 591> : Receive ACK Request

Received SIP PDU from ( 10.202.4.1:5060 )
INVITE sip:*7102@10.202.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-192e2717ef119f79-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10013@10.202.4.1:5060>
To: <sip:*7102@10.202.4.3:5060>
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=ff61ff1b
Call-ID: NmM4Y2VhMDcyMmNmZmMwNjA2YjJiNzJmZjg5YmRkNmY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 419

v=0
o=3cxPS 6593445888 454327009281 IN IP4 10.202.4.1
s=3cxPS Audio call
c=IN IP4 10.202.4.1
t=0 0
m=audio 7160 RTP/AVP 0 8 3 13 9 18 110 99 101
c=IN IP4 10.202.4.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 iLBC/8000
a=rtpmap:99 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv


Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-192e2717ef119f79-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=ff61ff1b
To: <sip:*7102@10.202.4.3:5060>
Call-ID: NmM4Y2VhMDcyMmNmZmMwNjA2YjJiNzJmZjg5YmRkNmY.
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


27 <Call 592> : ****** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337875319) ****
28 <SIP 592> : Receive INVITE Request
29 <NetCon 592> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1)
30 <NetCon 592> : Found inbound voip peer by dest-pattern id(12201)
31 <NetCon 592> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3)
32 <Call 592> : From Net - calledParty(*7102) callingParty(10013)
33 <Call 592> : MatchedPerfect
34 <Call 592> : MatchAllProcess After Sorted
<0> id(4588) dest(*....) prefer(0) selected(9)
<1> id(4589) dest(*....) prefer(0) selected(9)
<2> id(4590) dest(*....) prefer(0) selected(9)
<3> id(4591) dest(*....) prefer(0) selected(9)
35 <Call 592> : Initiate callee with dial-peer(*....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
36 <CEP 010000> : InitiateOutCall : calledNum(*7102), callingNum(10013), callerPort(ffffffff) type(GSM)
[166136.245] RTA(1/0/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1)
[166136.245] VP(1/0/0) open channel
[166136.245] VM(1/0/0) Tx GSM CallRequest 1 stage *7102
37 <CEP 010000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(592)
[166136.250] VM(1/0/0) set T38 enable by CCC
[166136.250] VM(1/0/0) set T38 mode STD
[166136.250] VM(1/0/0) Fax rate 9600
38 <SIP 592> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
39 <SIP 592> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
40 <PhonePlay 592> : Audio Count(1)
41 <PhonePlay 592> : rtpSessionId(1) Second Audio Port(-1)
42 <SIP 592> : SetAlerting
43 <Call 592> : PreConnected from(10000)
[166136.250] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[166136.250] VM(1/0/0) VAD disable
[166136.250] VP(1/0/0) update VAD 0
[166136.250] VM(1/0/0) SID enable by CCC
44 <SIP 592> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
45 <SIP 592> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10)
46 <SIP 592> : Add Local Audio MediaFormat : 0

Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-192e2717ef119f79-1---d8754z-;rport
From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=ff61ff1b
To: <sip:*7102@10.202.4.3:5060>;tag=774fa155a4
Call-ID: NmM4Y2VhMDcyMmNmZmMwNjA2YjJiNzJmZjg5YmRkNmY.
CSeq: 1 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:*7102@10.202.4.3
Content-Type: application/sdp
Content-Length: 240

v=0
o=*7102 1337875319 1337875319 IN IP4 10.202.4.3
s=AddPac Gateway SDP
c=IN IP4 10.202.4.3
t=1337875319 0
m=audio 24184 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

[166136.260] RTA(1/0/0) Rx RS_OPEN_REQ callId=592 ssId=1 G711U
peer=10.202.4.1 mp=24184/24185 hp=7160/7161
[166136.260] VM(1/0/0) codec same G711U
[166136.265] RTA(1/0/0) Rx RS_LISTEN_REQ callId=592 ssId=1 G711U
peer=10.202.4.1 mp=24184/24185 hp=7160/7161

Вот тут
viewtopic.php?f=11&t=2501
такую же проблему обсуждали, но вопрос переложили с больной головы на здоровую и благополучно забыли...