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mscheller

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Hallo ich habe einen Asterisk 1.4, und 2 Videotelefone Grandstream GXV3000
mein Problem ist, das keinen Videosignal übertragen wird. Meinen SIP.CONF

Code: 
[generall]
port = 5060
bindaddr = 0.0.0.0
context = sonstige
disallow = all
allow = ulaw
allow = alaw
allow = gsm
allow = g729
allow = h263
allow = h263p
allow = slinear
dtmfmode = rfc2833
canreinvite = no
videosupport=yes


[2000] ; Videophone
type = friend
context = mein-telefon
secret = 1234
host = dynamic


[8000]
type = friend
context = mein-telefon
secret = 1234
host = dynamic

[4400] ; Videophone
type = friend
context = mein-telefon
secret = 1234
host = dynamic

habe ich was vergessen?

Edit Guard-X: Code-Tags eingefügt!
 
Das GXV3000 unterstützt nur H.264 in älteren Softwareversionen...

mfg Guard-X
 
Danke für die schnelle Antwort, habe jetzt allow = h264 hinzugefügt leider bekomme ich immer noch kein Bild
 
habe ich eingestellt, bekomme aber immer noch kein Videobild..
 
Benutzt du zwei GXV3000 für deinen Test? Was sagt ein Debug?
 
ja 2 Stück... mein Debug? wie Starten in den .. -vvvvvc -> wenn so dann zeigt der nichts an auser den anruf auf das 2 Telefon
 
Funktioniert denn wenigstens die Verbindung zwischen diese beiden Geräten?

Zeig mal die Ausgabe von "sip debug peer xxx"
 
Zwischen beiden kann ich telefonieren.. Telefon1 2000 Telefon2 4400

Code: 
c=IN IP4 192.168.0.231
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20
m=video 5006 RTP/AVP 99
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0 KAFJWgUH5A,KM4BryA=
a=framerate:15

<------------->
--- (13 headers 18 lines) ---
Sending to 192.168.0.231 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]

<--- Reliably Transmitting (no NAT) to 192.168.0.231:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK66141006b9100126;received=192. 168.0.231
From: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
To: <sip:[email protected]>;tag=as17f0d4ac
Call-ID: [email protected]
CSeq: 5610 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dab411e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 m s (Method: INVITE)
Found user '4400'

<--- SIP read from 192.168.0.231:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bK66141006b9100126
From: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
To: <sip:[email protected]>;tag=as17f0d4ac
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 5610 ACK
User-Agent: Grandstream GXV3000 1.0.1.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0


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

<--- SIP read from 192.168.0.231:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bKd6b577e64853e995
From: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Supported: replaces, timer, path
Proxy-Authorization: Digest username="4400", realm="asterisk", algorithm=MD5, ur i="sip:[email protected]", nonce="2dab411e", response="b3387100c93d5ff742677fef e19b3cc5"
Call-ID: [email protected]
CSeq: 5611 INVITE
User-Agent: Grandstream GXV3000 1.0.1.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 432

v=0
o=4400 8000 8001 IN IP4 192.168.0.231
s=SIP Call
c=IN IP4 192.168.0.231
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20
m=video 5006 RTP/AVP 99
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0 KAFJWgUH5A,KM4BryA=
a=framerate:15

<------------->
--- (14 headers 18 lines) ---
Sending to 192.168.0.231 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found user '4400'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 3
Found RTP video format 99
Peer audio RTP is at port 192.168.0.231:5004
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format GSM for ID 3
Found unknown media description format H264 for ID 99
Capabilities: us - 0x38000e (gsm|ulaw|alaw|h263|h263p|h264), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing),  combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.231:5004
Looking for 2000 in mein-telefon (domain 192.168.0.211)
list_route: hop: <sip:[email protected]:5060>

<--- Transmitting (no NAT) to 192.168.0.231:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bKd6b577e64853e995;received=192. 168.0.231
From: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 5611 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
    -- Executing [2000@mein-telefon:1] Dial("SIP/4400-081d0ab8", "SIP/2000") in new stack
    -- Called 2000
    -- SIP/2000-081d2048 is ringing

<--- Transmitting (no NAT) to 192.168.0.231:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bKd6b577e64853e995;received=192. 168.0.231
From: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
To: <sip:[email protected]>;tag=as5795b2cf
Call-ID: [email protected]
CSeq: 5611 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
    -- SIP/2000-081d2048 answered SIP/4400-081d0ab8
Audio is at 192.168.0.211 port 11678
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bKd6b577e64853e995;received=192. 168.0.231
From: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
To: <sip:[email protected]>;tag=as5795b2cf
Call-ID: [email protected]
CSeq: 5611 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 231

v=0
o=root 4181 4181 IN IP4 192.168.0.211
s=session
c=IN IP4 192.168.0.211
t=0 0
m=audio 11678 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Native bridging SIP/4400-081d0ab8 and SIP/2000-081d2048

<--- SIP read from 192.168.0.231:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.231:5060;branch=z9hG4bKd693dae5530599d1
From: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
To: <sip:[email protected]>;tag=as5795b2cf
Contact: <sip:[email protected]:5060>
Proxy-Authorization: Digest username="4400", realm="asterisk", algorithm=MD5, ur i="sip:[email protected]", nonce="2dab411e", response="b3387100c93d5ff742677fef e19b3cc5"
Call-ID: [email protected]
CSeq: 5611 ACK
User-Agent: Grandstream GXV3000 1.0.1.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.231, port 5060
Audio is at 192.168.0.211 port 11678
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.0.231:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK1ee4943b;rport
From: <sip:[email protected]>;tag=as5795b2cf
To: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 183

v=0
o=root 4181 4182 IN IP4 192.168.0.230
s=session
c=IN IP4 192.168.0.230
t=0 0
m=audio 5004 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from 192.168.0.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK1ee4943b;rport
From: <sip:[email protected]>;tag=as5795b2cf
To: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Grandstream GXV3000 1.0.1.27
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Supported: replaces, timer, 100rel, path
Content-Length: 157

v=0
o=4400 8000 8002 IN IP4 192.168.0.231
s=SIP Call
c=IN IP4 192.168.0.231
t=0 0
m=audio 5004 RTP/AVP 0
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 192.168.0.231:5004
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), comb ined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing),  combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.231:5004
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.231, port 5060
Transmitting (no NAT) to 192.168.0.231:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK047a1751;rport
From: <sip:[email protected]>;tag=as5795b2cf
To: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.231, port 5060
Audio is at 192.168.0.211 port 11678
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.0.231:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK5bd19526;rport
From: <sip:[email protected]>;tag=as5795b2cf
To: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 184

v=0
o=root 4181 4183 IN IP4 192.168.0.211
s=session
c=IN IP4 192.168.0.211
t=0 0
m=audio 11678 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
  == Spawn extension (mein-telefon, 2000, 1) exited non-zero on 'SIP/4400-081d0a b8'

<--- SIP read from 192.168.0.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK5bd19526;rport
From: <sip:[email protected]>;tag=as5795b2cf
To: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Grandstream GXV3000 1.0.1.27
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Supported: replaces, timer, 100rel, path
Content-Length: 157

v=0
o=4400 8000 8003 IN IP4 192.168.0.231
s=SIP Call
c=IN IP4 192.168.0.231
t=0 0
m=audio 5004 RTP/AVP 0
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 m s (Method: ACK)
Found RTP audio format 0
Peer audio RTP is at port 192.168.0.231:5004
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), comb ined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing),  combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.231:5004
list_route: hop: <sip:[email protected]:5060>
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.231, port 5060
Transmitting (no NAT) to 192.168.0.231:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK4f629fea;rport
From: <sip:[email protected]>;tag=as5795b2cf
To: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.231, port 5060
Reliably Transmitting (no NAT) to 192.168.0.231:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK63c4ed9e;rport
From: <sip:[email protected]>;tag=as5795b2cf
To: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
Call-ID: [email protected]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]' in 32000 m s (Method: ACK)

<--- SIP read from 192.168.0.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK63c4ed9e;rport
From: <sip:[email protected]>;tag=as5795b2cf
To: "4400" <sip:[email protected]>;tag=9eb1f1f1c434d5d1
Call-ID: [email protected]
CSeq: 104 BYE
User-Agent: Grandstream GXV3000 1.0.1.27
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer, 100rel, path
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK
Reliably Transmitting (no NAT) to 192.168.0.231:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK002e519c;rport
From: "asterisk" <sip:[email protected]>;tag=as447dc581
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 27 Feb 2008 12:36:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---

<--- SIP read from 192.168.0.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK002e519c;rport
From: "asterisk" <sip:[email protected]>;tag=as447dc581
To: <sip:[email protected]:5060>;tag=72058880891080b0
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Grandstream GXV3000 1.0.1.27
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer, 100rel, path
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Me thod: OPTIONS
Really destroying SIP dialog '[email protected]' Method: REGISTER
Reliably Transmitting (no NAT) to 192.168.0.231:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK76a527ba;rport
From: "asterisk" <sip:[email protected]>;tag=as6b0bc7c8
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 27 Feb 2008 12:37:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---

<--- SIP read from 192.168.0.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK76a527ba;rport
From: "asterisk" <sip:[email protected]>;tag=as6b0bc7c8
To: <sip:[email protected]:5060>;tag=dab2207720135703
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Grandstream GXV3000 1.0.1.27
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer, 100rel, path
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: OPTIONS

Edit Guard-X: Code-Tags eingefügt!
 
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