1.5.19 поставил, видео не появилось, зато в логе увидел следующее:

[08-01-19]19:45:22.882 | Error | CCM | "Failed to create or negotiate SDP contents. [300@192.168.1.251 -> 301@192.168.1.251]" | sua::CSIPCall::OnMediaSessionClosed
[08-01-19]19:45:22.882 | Info | CCM | "[EConnected|ESDPAnswered|F][containee:T] [300@192.168.1.251 -> 301@192.168.1.251]" | sua::CSIPCall::InternalShutdown
[08-01-19]19:45:22.882 | Info | RESIP:DUM | "InviteSession::Connected: end" |
[08-01-19]19:45:22.882 | Info | RESIP:DUM | """Admin""<sip:300@192.168.1.251>;tag=040ed452 Sending BYE Application Rejected Sdp(usually no common codec)" |
[08-01-19]19:45:22.882 | Info | RESIP:DUM | "Transition InviteSession::Connected -> InviteSession::Terminated" |
[08-01-19]19:45:22.882 | Info | CCM | "[ECallHangingUp|ESDPAnswered|T][containee:T] [300@192.168.1.251 -> 301@192.168.1.251]" | sua::CSIPCall::InternalShutdown
[08-01-19]19:45:22.887 | Info | AbstractPhone | "Hangup" | AbPhone::CCall::Hangup
[08-01-19]19:45:22.887 | Info | AbstractPhone | "Cancel" | AbPhone::COutgoingCallRequest::Cancel
[08-01-19]19:45:22.887 | Info | Media | " [session-ID:711]" | sua::CMediaSession::~CMediaSession
[08-01-19]19:45:22.887 | Info | CCM | "[ECallHungUp|ESDPAnswered|T][containee:T] [300@192.168.1.251 -> 301@192.168.1.251]" | sua::CSIPCall::InternalShutdown
[08-01-19]19:45:22.887 | Info | CCM | "Object already shut-down. Waiting for deletion/final unreference only. [300@192.168.1.251 -> 301@192.168.1.251]" | sua::CSIPOutgoingCallGroup::OnMediaSessionClosed
[08-01-19]19:45:22.887 | Error | AbstractPhone | "OnOutgoingCallRequestTerminated" | AbPhone::COutgoingCallRequest::CInviteSessionSink::BreakCircularReference
[08-01-19]19:45:22.887 | Info | Audio | "StopAudio entry point, where instance ID=643" | sua::CAudioManager::StopAudio
[08-01-19]19:45:22.889 | Info | Media | " [session-ID:672]" | sua::CMediaSession::~CMediaSession
[08-01-19]19:45:22.985 | Info | RESIP:DUM | "Got: SipResp: 200 tid=83479f07905cca1a cseq=BYE contact=301@192.168.1.251 / 4 from(wire)" |
[08-01-19]19:45:22.988 | Info | RESIP:DUM | "InviteSession::dispatchTerminated SipResp: 200 tid=83479f07905cca1a cseq=BYE contact=301@192.168.1.251 / 4 from(wire)" |
[08-01-19]19:45:29.242 | Info | USB Devices | "First attempt to connect device" | cpsi::CPolycomUSB::Connect
[08-01-19]19:45:29.245 | Error | USB Devices | "Failed to create CLSID_PlcmGobiCtrl instance" | cpsi::CPolycomUSB::Connect
[08-01-19]19:45:32.945 | Info | RESIP:TRANSACTION | "Stats message " |
[08-01-19]19:45:32.945 | Warning | RESIP:STATS | "RESIP:TRANSACTION
TU summary: 0 TRANSPORT 0 TRANSACTION 0 CLIENTTX 5 SERVERTX 0 TIMERS 9
Transaction summary: reqi 2 reqo 27 rspi 31 rspo 2
Details: INVi 0/S0/F0 INVo 7/S4/F3 ACKi 0 ACKo 7 BYEi 2/S2/F0 BYEo 1/S1/F0 CANi 0/S2/F0 CANo 0/S0/F0 MSGi 0/S0/F0 MSGo 0/S0/F0 OPTi 0/S0/F0 OPTo 0/S0/F0 REGi 0/S0/F0 REGo 2/S1/F1 PUBi 0/S0/F0 PUBo 0/S0/F0 SUBi 0/S0/F0 SUBo 10/S0/F10 NOTi 0/S0/F0 NOTo 0/S0/F0
Retransmissions: INVx 0 BYEx 0 CANx 0 MSGx 0 OPTx 0 REGx 0 finx 0 nonx 0 PUBx 0 SUBx 0 NOTx 0" |
[08-01-19]19:45:34.457 | Info | RESIP:DUM | "Got: SipResp: 401 tid=dd68f67a3c60c178 cseq=SUBSCRIBE / 1 from(wire)" |
[08-01-19]19:45:34.559 | Info | RESIP:DUM | "Got: SipResp: 404 tid=19456d4d2836035c cseq=SUBSCRIBE / 2 from(wire)" |
[08-01-19]19:45:34.559 | Info | CCM | "Will attempt to re-SUBSCRIBE in 180 sec.[300@192.168.1.251 -> 300@192.168.1.251]" | sua::CSIPClientSubscriptionWatcher::OnSIPClientSubscriptionSessionDestroyed
[08-01-19]19:45:36.182 | Info | USB Devices | "First attempt to connect device" | cpsi::CPolycomUSB::Connect
[08-01-19]19:45:36.182 | Error | USB Devices | "Failed to create CLSID_PlcmGobiCtrl instance" | cpsi::CPolycomUSB::Connect

Не обращайте вимание на дату и время, это у меня на ноуте они неправильные.

Подскажите если кто зает, как с этим бороться, возможно лучше обратиться на форум Логитека, или eyeBeam'а, но можт тут кто сталкивался... Самое интересное что сам себя я прекрасно вижу, ошибка выскакивает при попытке стартовать видео и звонок разрывается.

Что-то не понятно кто с кем соединяется: [300@192.168.1.251 -> 301@192.168.1.251]
300 с 301 на том же ИП адресе 192.168.1.251?

Нет, 192.168.1.251 это астериск, а 300 и 301 это два пира, которые сидят на 192.168.1.100 и 192.168.1.13 соответственно. Просто в поле domain настроек клиента у меня написан IP-адрес астериска, а не доменное имя.

Сейчас поправил, написал в sip.conf domain=ves_lan и клиентам в настройках домен ves_lan. Никаких положительных тенденций не обнаружено.

Вот например недвусмысленное сообщение в самом начале
Sending BYE Application Rejected Sdp(usually no common codec)"
Дальше можно уже не ломать глаза.

Оставил на клиентах из видео-кодеков только H.263p. При этом имеем в дебаге:

<--- SIP read from 192.168.1.71:27022 --->
INVITE sip:301@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.71:27022;branch=z9hG4bK-d8754z-500ed07c52056629-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:300@192.168.1.71:27022>
To: "301"<sip:301@ves_lan>;tag=as3699903a
From: "Admin"<sip:300@ves_lan>;tag=4452221e
Call-ID: NjIxZjRlZGYzNThhMzc2ZjkyZDQ5NjlkZTU4OTViZjg.
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="300",realm="ves_lan",nonce="25062c67",uri="sip:301@192.168.1.251",response="2b572bd587f55aa17f7d65292a6ba019",algorithm=MD5
User-Agent: eyeBeam release 1100z stamp 47739
Content-Length: 509

v=0
o=- 2 3 IN IP4 192.168.1.71
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.71
t=0 0
m=audio 7316 RTP/AVP 0 8 3 101
a=alt:1 1 : xMzxHNvK mZAVJ5hP 192.168.1.71 7316
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:9063D431EF08411FBC75C5F2714AA491
m=video 55858 RTP/AVP 115
a=alt:1 1 : uoEJaKLI JyMKt3xM 192.168.1.71 55858
a=fmtp:115 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520
a=rtpmap:115 H263-1998/90000
a=sendrecv
a=x-rtp-session-id:CC8E655D72654697B08F2C4C906CEA2B

<------------->
--- (13 headers 17 lines) ---
Sending to 192.168.1.71 : 27022 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found RTP video format 115
Peer audio RTP is at port 192.168.1.71:7316
Found audio description format telephone-event for ID 101
Found video description format H263-1998 for ID 115
Capabilities: us - 0x3c000e (gsm|ulaw|alaw|h261|h263|h263p|h264), peer - audio=0x10000e (gsm|ulaw|alaw|h263p)/video=0x100000 (h263p), combined - (ulaw|alaw|gsm|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.71:7316
Peer video RTP is at port 192.168.1.71:55858

<--- Transmitting (NAT) to 192.168.1.71:27022 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.71:27022;branch=z9hG4bK-d8754z-500ed07c52056629-1---d8754z-;rport;received=192.168.1.71
From: "Admin"<sip:300@ves_lan>;tag=4452221e
To: "301"<sip:301@ves_lan>;tag=as3699903a
Call-ID: NjIxZjRlZGYzNThhMzc2ZjkyZDQ5NjlkZTU4OTViZjg.
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.251>
Content-Length: 0


<------------>
Video is at 192.168.1.251 port 16284
Audio is at 192.168.1.251 port 14786
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.1.71:27022 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.71:27022;branch=z9hG4bK-d8754z-500ed07c52056629-1---d8754z-;rport;received=192.168.1.71
From: "Admin"<sip:300@ves_lan>;tag=4452221e
To: "301"<sip:301@ves_lan>;tag=as3699903a
Call-ID: NjIxZjRlZGYzNThhMzc2ZjkyZDQ5NjlkZTU4OTViZjg.
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.251>
Content-Type: application/sdp
Content-Length: 368

v=0
o=root 35894 35896 IN IP4 192.168.1.251
s=session
c=IN IP4 192.168.1.251
b=CT:1024
t=0 0
m=audio 14786 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
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 16284 RTP/AVP 103
a=rtpmap:103 h263-1998/8000
a=sendrecv

<------------>
helios*CLI>
<--- SIP read from 192.168.1.71:27022 --->
ACK sip:301@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.71:27022;branch=z9hG4bK-d8754z-211de532aa65712c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:300@192.168.1.71:27022>
To: "301"<sip:301@ves_lan>;tag=as3699903a
From: "Admin"<sip:300@ves_lan>;tag=4452221e
Call-ID: NjIxZjRlZGYzNThhMzc2ZjkyZDQ5NjlkZTU4OTViZjg.
CSeq: 3 ACK
Proxy-Authorization: Digest username="300",realm="ves_lan",nonce="25062c67",uri="sip:301@192.168.1.251",response="2b572bd587f55aa17f7d65292a6ba019",algorithm=MD5
User-Agent: eyeBeam release 1100z stamp 47739
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---

<--- SIP read from 192.168.1.71:27022 --->
BYE sip:301@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.71:27022;branch=z9hG4bK-d8754z-19507d7b7a731d31-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:300@192.168.1.71:27022>
To: "301"<sip:301@ves_lan>;tag=as3699903a
From: "Admin"<sip:300@ves_lan>;tag=4452221e
Call-ID: NjIxZjRlZGYzNThhMzc2ZjkyZDQ5NjlkZTU4OTViZjg.
CSeq: 4 BYE
Proxy-Authorization: Digest username="300",realm="ves_lan",nonce="25062c67",uri="sip:301@192.168.1.251",response="d35bfef7304b8e4d522fa0c4d45ff4d6",algorithm=MD5
User-Agent: eyeBeam release 1100z stamp 47739
Reason: SIP;description="Application Rejected Sdp(usually no common codec)"
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.1.71 : 27022 (NAT)

<--- Transmitting (NAT) to 192.168.1.71:27022 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.71:27022;branch=z9hG4bK-d8754z-19507d7b7a731d31-1---d8754z-;rport;received=192.168.1.71
From: "Admin"<sip:300@ves_lan>;tag=4452221e
To: "301"<sip:301@ves_lan>;tag=as3699903a
Call-ID: NjIxZjRlZGYzNThhMzc2ZjkyZDQ5NjlkZTU4OTViZjg.
CSeq: 4 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.251>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3f61018074a3072c27908bbd7359b98c@192.168.1.251' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:301@192.168.1.13:41606;rinstance=d778d375e946e37f> for address/port to send to
set_destination: set destination to 192.168.1.13, port 41606
Reliably Transmitting (no NAT) to 192.168.1.13:41606:
BYE sip:301@192.168.1.13:41606;rinstance=d778d375e946e37f SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK7db97148
From: "Admin" <sip:300@192.168.1.251>;tag=as7e85bc45
To: <sip:301@192.168.1.13:41606;rinstance=d778d375e946e37f>;tag=943c8704
Call-ID: 3f61018074a3072c27908bbd7359b98c@192.168.1.251
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
helios*CLI>
<--- SIP read from 192.168.1.13:41606 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK7db97148
Contact: <sip:301@192.168.1.13:41606;rinstance=d778d375e946e37f>
To: <sip:301@192.168.1.13:41606;rinstance=d778d375e946e37f>;tag=943c8704
From: "Admin"<sip:300@192.168.1.251>;tag=as7e85bc45
Call-ID: 3f61018074a3072c27908bbd7359b98c@192.168.1.251
CSeq: 103 BYE
User-Agent: eyeBeam release 1100z stamp 47739
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '3f61018074a3072c27908bbd7359b98c@192.168.1.251' Method: INVITE
Really destroying SIP dialog 'NjIxZjRlZGYzNThhMzc2ZjkyZDQ5NjlkZTU4OTViZjg.' Method: BYE
helios*CLI>
<--- SIP read from 192.168.1.13:41606 --->



<------------->
--- (0 headers 1 lines) --

А на клиенте в момент старта видео:
[08-01-20]00:27:58.975 | Info | USB Devices | "First attempt to connect device" | cpsi::CPolycomUSB::Connect
[08-01-20]00:27:58.978 | Error | USB Devices | "Failed to create CLSID_PlcmGobiCtrl instance" | cpsi::CPolycomUSB::Connect

Уважыемый bonnyfacy, не надо дебаг-портянки такого размера сюда постить. Вы это делаете для себя, там и пытайтесь искать зерно истниы. Глаза сломать можно! зачем тут такая избыточная инфа -

---
helios*CLI>
<--- SIP read from 192.168.1.13:41606 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK7db97148
Contact: <sip:301@192.168.1.13:41606;rinstance=d778d375e946e37f>
To: <sip:301@192.168.1.13:41606;rinstance=d778d375e946e37f>;tag=943c8704
From: "Admin"<sip:300@192.168.1.251>;tag=as7e85bc45
Call-ID: 3f61018074a3072c27908bbd7359b98c@192.168.1.251
CSeq: 103 BYE
User-Agent: eyeBeam release 1100z stamp 47739
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '3f61018074a3072c27908bbd7359b98c@192.168.1.251' Method: INVITE
Really destroying SIP dialog 'NjIxZjRlZGYzNThhMzc2ZjkyZDQ5NjlkZTU4OTViZjg.' Method: BYE
helios*CLI>
<--- SIP read from 192.168.1.13:41606 --->

Из всей этой кучи ясно и понятно:

CSeq: 4 BYE
Proxy-Authorization: Digest username="300",realm="ves_lan",nonce="25062c67",uri="sip:301@192.168.1.251",response="d35bfef7304b8e4d522fa0c4d45ff4d6",algorithm=MD5
User-Agent: eyeBeam release 1100z stamp 47739
Reason: SIP;description="Application Rejected Sdp(usually no common codec)"

Может у вас PolycomUSB болеет, может высокое давление, или кого арестовали за взятки? Изучайте мат часть:
CLI>show file formats
Format Name Extensions
g729 g729 g729
h263 h263 h263
gsm gsm gsm
slin sln sln|raw
adpcm vox vox
h264 h264 h264
g726 g726-16 g726-16
g726 g726-24 g726-24
g726 g726-32 g726-32
g726 g726-40 g726-40
g723 g723sf g723|g723sf
gsm wav49 WAV|wav49
ilbc iLBC ilbc
slin ogg_vorbis ogg
slin mp3 mp3
g722 g722 g722
ulaw au au
alaw alaw alaw|al
ulaw pcm pcm|ulaw|ul|mu
slin wav wav
Сделайте сначала простую схему: eyeBeam <-> eyeBeam или Ekiga <-> Ekiga
Если работает - хорошо, а нет, так оцените: сколько усилий Вы готовы положить, чтобы заработало? Оно нужно ли?

Я бы предложил использовать любой pastebin для логов, например, http://pastebin.ca.
Мы давно думаем о прикрутке внутреннего пастебина, но, как обычно, руки не доходят :-)