Afficher flux video d'une caméra dans Xlite

[greg2901]

Bonjour,

J'ai déconnecté tous les postes sauf le SPA 3102 qui visiblement ne pollue pas trop le log, et voici tout ce que j'ai pu copier

From: <sip:5520@192.168.1.220>;tag=as76046a24
To: "5501"<sip:5501@192.168.1.220>;tag=e6894cea
Contact: <sip:5520@192.168.1.220:5060>
Call-ID: YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
set_destination: set destination to 192.168.1.151:5060
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.1.151:5060:
INVITE sip:5520@192.168.1.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK36ef104f
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.6.0
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: 202

v=0
o=root 752446342 752446344 IN IP4 192.168.1.11
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.1.11
t=0 0
m=audio 59984 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.151:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK36ef104f
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 104 INVITE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.151:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK36ef104f
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 104 INVITE
Contact: <sip:5520@192.168.1.151:5060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Type: application/sdp
Content-Length: 179

v=0
o=userX 20000001 20000001 IN IP4 192.168.1.151
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.1.151
t=0 0
m=audio 10500 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
<------------->
--- (10 headers 8 lines) ---
set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
set_destination: set destination to 192.168.1.151:5060
Transmitting (no NAT) to 192.168.1.151:5060:
ACK sip:5520@192.168.1.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK7ee8a187
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.11:45996 --->
BYE sip:5520@192.168.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45996;branch=z9hG4bK-d8754z-2173cd01d15af29a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5501@192.168.1.11:45996;transport=udp>
To: <sip:5520@192.168.1.220>;tag=as76046a24
From: "5501"<sip:5501@192.168.1.220>;tag=e6894cea
Call-ID: YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.
CSeq: 3 BYE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="5501",realm="asterisk",nonce="75510cc6", uri="sip:5520@192.168.1.220:5060",response="3e6e4c 61dbb314c9ae372b80e3f83b20",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.11:45996 (no NAT)
Scheduling destruction of SIP dialog 'YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.1.11:45996 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:45996;branch=z9hG4bK-d8754z-2173cd01d15af29a-1---d8754z-;received=192.168.1.11;rport=45996
From: "5501"<sip:5501@192.168.1.220>;tag=e6894cea
To: <sip:5520@192.168.1.220>;tag=as76046a24
Call-ID: YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.
CSeq: 3 BYE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
set_destination: set destination to 192.168.1.151:5060
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.1.151:5060:
INVITE sip:5520@192.168.1.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK12711fef
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 105 INVITE
User-Agent: Asterisk PBX 1.8.6.0
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: 204

v=0
o=root 752446342 752446345 IN IP4 192.168.1.220
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.1.220
t=0 0
m=audio 11108 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.151:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK12711fef
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 105 INVITE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.151:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK12711fef
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 105 INVITE
Contact: <sip:5520@192.168.1.151:5060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Type: application/sdp
Content-Length: 179

v=0
o=userX 20000001 20000001 IN IP4 192.168.1.151
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.1.151
t=0 0
m=audio 10500 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
<------------->
--- (10 headers 8 lines) ---
set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
set_destination: set destination to 192.168.1.151:5060
Transmitting (no NAT) to 192.168.1.151:5060:
ACK sip:5520@192.168.1.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK5ab0f8d3
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 105 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:50 60' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
set_destination: set destination to 192.168.1.151:5060
Reliably Transmitting (no NAT) to 192.168.1.151:5060:
BYE sip:5520@192.168.1.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK19c8feec
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 106 BYE
User-Agent: Asterisk PBX 1.8.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
== Spawn extension (phones, 5520, 2) exited non-zero on 'SIP/5501-0000002c'
== Extension Changed 5501[subscriptions] new state Idle for Notify User 5503 (queued)

<--- SIP read from UDP:192.168.1.151:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK19c8feec
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 106 BYE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.151:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK19c8feec
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 106 BYE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:50 60' Method: INVITE

[therebel23]

Apparemment tu appelles avec X-Lite, on voit que X-Lite propose du G711u et G711a.

Une fois que tu as décroché, est-ce que tu actives bien la video en affichant l'onglet video dans X-Lite ? Parce qu'il n'y a aucune trace de codec video dans le log que tu nous donnes ..

[greg2901]

Oui j'utilise même X lite 4 et il y a une fonction qui permet de faire un appel vidéo direct, du coup j'ai installé x lite sur un autre PC et la j'arrive a avoir la vidéo. Je pense que mon problème vient de la camera, je vais lui faire un petit retour usine et reprendre la configuration depuis le début j'ai probablement loupé quelque chose...