Voir la version complète : Afficher flux video d'une caméra dans Xlite
greg2901
14/11/2011, 22h44
Bonjour,
Je dispose d'une camera IP Mobotix, elle offre la particularité de gérer un compte sip et se comporte comme un téléphone. Je souhaiterais afficher la vidéo avec Xilte (d’après la doc c'est possible), j'ai tout bien paramétré dans sip.conf:
videosupport=yes
allow=h263
Dans xlite j'ai gardé que le codec H263 (le seule dispo dans la camera).
Lors d'un appel, la camera répond, j'ai la phonie mais de vidéo et au niveau CLI j'ai ca:
-- Executing [5520@phones:1] Set("SIP/5501-0000000a", "CHANNEL(language)=fr") in new stack
-- Executing [5520@phones:2] Dial("SIP/5501-0000000a", "SIP/5520,20") in new stack
== Extension Changed 5501[subscriptions] new state InUse for Notify User 5503
== Using SIP RTP CoS mark 5
-- Called SIP/5520
-- SIP/5520-0000000b is ringing
-- SIP/5520-0000000b answered SIP/5501-0000000a
-- Remotely bridging SIP/5501-0000000a and SIP/5520-0000000b
[Nov 14 21:51:34] WARNING[3087]: chan_sip.c:8738 process_sdp: Unsupported SDP media type in offer: video 0 RTP/AVP 34
== Spawn extension (phones, 5520, 2) exited non-zero on 'SIP/5501-0000000a'
== Extension Changed 5501[subscriptions] new state Idle for Notify User 5503
et pas de vidéo....
Une idée ?
Merci
La meme tace avec en plus les debug sip activés, ca serait plus pratique pour comprendre:
sip set debug on
puis relancer le test
greg2901
15/11/2011, 20h57
Voila le résultat, je me connecte en ssh avec putty c'est le max que je puisse extraire:
--- (14 headers 0 lines) ---
Sending to 192.168.1.41:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.1.41:5060:
OPTIONS sip:5503@192.168.1.41:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK3adf2a63
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.220>;tag=as7ac5f647
To: <sip:5503@192.168.1.41:5060;transport=udp>
Contact: <sip:asterisk@192.168.1.220:5060>
Call-ID: 12cf549f6f453b9b134b5da37a7db21d@192.168.1.220:506 0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.6.0
Date: Tue, 15 Nov 2011 19:04:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to 192.168.1.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.41;branch=z9hG4bK843c192f58aaf3a64;recei ved=192.168.1.41
From: <sip:5503@192.168.1.220:5060>;tag=09aa42ea0c
To: <sip:5503@192.168.1.220:5060>;tag=as042c4e18
Call-ID: bf0fa8af85b3dda3
CSeq: 100984 REGISTER
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: <sip:5503@192.168.1.41:5060;transport=udp>;expires=120
Date: Tue, 15 Nov 2011 19:04:36 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'bf0fa8af85b3dda3' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.1.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK3adf2a63
From: "asterisk" <sip:asterisk@192.168.1.220>;tag=as7ac5f647
To: <sip:5503@192.168.1.41:5060;transport=udp>;tag=3621404576
Call-ID: 12cf549f6f453b9b134b5da37a7db21d@192.168.1.220:506 0
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 55i/3.2.0.1011
Supported: path
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '12cf549f6f453b9b134b5da37a7db21d@192.168.1.220:50 60' Method: OPTIONS
<--- SIP read from UDP:192.168.1.11:45688 --->
BYE sip:5520@192.168.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45688;branch=z9hG4bK-d8754z-d464ac1d05087289-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5501@192.168.1.11:45688;transport=udp>
To: <sip:5520@192.168.1.220>;tag=as1606e0bb
From: "5501"<sip:5501@192.168.1.220>;tag=ce578432
Call-ID: MmQxNGNiMGEwYjFiZDM2MjQ3MzViNjA3ZDM3MTEwMGY.
CSeq: 3 BYE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="5501",realm="asterisk",nonce="61f12305",uri="sip:5520@192.168.1.220:5060",response="184bcc94cc53ebec83743cb0860427d7",algorithm=MD5
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.11:45688 (no NAT)
Scheduling destruction of SIP dialog 'MmQxNGNiMGEwYjFiZDM2MjQ3MzViNjA3ZDM3MTEwMGY.' in 32000 ms (Method: BYE)
<--- Transmitting (no NAT) to 192.168.1.11:45688 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:45688;branch=z9hG4bK-d8754z-d464ac1d05087289-1---d8754z-;received=192.168.1.11;rport=45688
From: "5501"<sip:5501@192.168.1.220>;tag=ce578432
To: <sip:5520@192.168.1.220>;tag=as1606e0bb
Call-ID: MmQxNGNiMGEwYjFiZDM2MjQ3MzViNjA3ZDM3MTEwMGY.
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=z9hG4bK1f095ee8
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 219adec357d78ce34c2273335129e215@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: 203
v=0
o=root 993097821 993097824 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 7872 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=z9hG4bK1f095ee8
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Call-ID: 219adec357d78ce34c2273335129e215@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=z9hG4bK1f095ee8
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Call-ID: 219adec357d78ce34c2273335129e215@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=z9hG4bK49f26b0c
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 219adec357d78ce34c2273335129e215@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 '219adec357d78ce34c2273335129e215@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=z9hG4bK33cc2e46
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Call-ID: 219adec357d78ce34c2273335129e215@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/5501-00000019'
set_destination: Parsing <sip:5503@192.168.1.41:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.1.41:5060
Reliably Transmitting (no NAT) to 192.168.1.41:5060:
NOTIFY sip:5503@192.168.1.41:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK73d30871
Max-Forwards: 70
From: <sip:5501@192.168.1.220:5060>;tag=as77c8c620
To: <sip:5503@192.168.1.220:5060>;tag=560a44aa4f
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 4baa7bb7f47d8d4c
Seq: 164 NOTIFY
User-Agent: Asterisk PBX 1.8.6.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 209
<?xml version="1.0"?><dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="62" state="full" entity="sip:5501@192.168.1.220:5060"><dialog id="5501"><state>terminated</state>
</dialog>
</dialog-info>
---
== Extension Changed 5501[subscriptions] new state Idle for Notify User 5503
<--- 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=z9hG4bK33cc2e46
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Call-ID: 219adec357d78ce34c2273335129e215@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=z9hG4bK33cc2e46
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Call-ID: 219adec357d78ce34c2273335129e215@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 '219adec357d78ce34c2273335129e215@192.168.1.220:50 60' Method: INVITE
<--- SIP read from UDP:192.168.1.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK73d30871
From: <sip:5501@192.168.1.220:5060>;tag=as77c8c620
To: <sip:5503@192.168.1.220:5060>;tag=560a44aa4f
Call-ID: 4baa7bb7f47d8d4c
CSeq: 164 NOTIFY
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Server: Aastra 55i/3.2.0.1011
Supported: path
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
serveur*CLI> sip set debug off
SIP Debugging Disabled
C'est un peu obscure pour moi, si ça vous parle n’hésitez pas a m'éclairer!!!!
Merci
therebel23
16/11/2011, 21h38
Il faudrait que tu débranches ton aastra 55i au moment ou tu fais ton test de visio entre ta caméra et ton PC.. et re-poster le résultat ..
greg2901
20/11/2011, 00h27
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="3e6e4c61dbb314c9ae372b80e3f83b20",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
20/11/2011, 00h39
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
20/11/2011, 10h59
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...
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