VTO3211D + VTH-2421FW-P Asterisk Sip (Kein Video im Call VTO->VTH))

[fluffymadness]

edit: Fehlerhafte extensions.conf überarbeitet.
Neuer Sip-Debuglog für fehlerhaften Anruf ohne Video bis zum Disconnect

Hallo erstmal.
Mitlerweile bin ich auch Besitzer von Dahua Equipment.
Ich hatte nach einigen Tagen Konfiguration das VTO am laufen mit ASTERISK

Was funktionierte:

- Beim betätigen der Glocke, haben alle Sip Telefone an der Fritzbox geläutet, Die Linphone Clients am Handy,
Jitsi am PC. Beim Aufbauen der Verbindung gab es Video von der Glocke an Jitsi und Linphone.
Soweit so gut.

Das Grauen fing erst an nachdem auch das VTH per Post kam.
Nach stundenlangen Konfigurieren ging es ohne Probleme das ich vom VTH aus einen Linephone Client anrufen konnte (Video + Audio),
auch umgekehrt ging ohne Probleme, Linphone zu VTH mit (Video + Audio). Ich konnte auch vom VTH aus raustelefonieren über unseren VOIP Provider.

Das was natürlich nicht geht ist ein normaler Anruf von der Glocke zum VTH.
Es kommt nur Ton an und kein Video.
Auch umgekehrt vom VTH den Anruf auf die Glocke zu initieren (via Monitor Bereich am VTH)(Glocke hebt automatisch ab) liefert mir nur Ton
und beendet mir den Anruf nach ein paar Sekunden mit einem Network Abnormal Video Connection Timeout.

Das Video vom VTO seh ich am VTH beim Anruf, da es ja über einen anderen Kanal geht
und erst beim Abheben das Ding auf Sip-Video umschaltet:
Verbindung zwischen VTH und VTO funktioniert ist also korrekt eingestellt.

Mittlerweile habe ich meine Asterisk Configuration auf ein Minimum verringert. Das Sip Debug Log spuckt zwar einiges aus
aber warum diese Probleme auftreten ist nicht gerade verständlich.
Ich hab unten mal meine komplett abgespeckte config angehängt vielleicht kann mir ja irgendwer helfen.
Vielen Dank schonmal im Vorraus.



Hier die Daten des Setups:

Asterisk 16.2.1
IP: 192.168.1.219

VTO-3211D
UserAgent bei Asterisk: VTO3221D V4.300.0.5
IP: 192.168.1.13

VTH-2421FW-P
UserAgent bei Asterisk: VTH2421F_R V4.410.0.4
IP: 192.168.1.245

sip.conf
[general]
transport=udp
bindport=5060
bindaddr=0.0.0.0
localnet=192.168.1.0/255.255.255.0 ;change this according to your LAN configuration
nat=force_rport,comedia instead
language=en
srvlookup=no
disallow=all
dateformat=%F %T
alwaysauthreject=yes            ; Wir lassen abgewiesene User nicht wissen DAS es diesen User mit falschem Pwd auch wirklich gibt..!
allowguest=no                       ; Wir verbieten "Guest" Registrationen, also ohne definierte Anmeldedaten unsererseits.
tcpenable=yes
videosupport=yes
directmedia=yes
srvlookup=yes
;prematuremedia=no
;progressinband=no


[204] ; internal extension ; SIP phones must be connected to LAN
user=204
type=friend
secret=204204
host=dynamic
qualify=yes
context=from-internal
disallow=all
allow=ulaw
allow=alaw
allow=h264
dtmfmode=info
canreinvite=yes
insecure=invite,port

[8001] ; internal extension ; SIP phones must be connected to LAN
user=8001
type=friend
secret=80018001
host=dynamic
qualify=yes
context=from-internal
disallow=all
allow=ulaw
allow=alaw
allow=h264
dtmfmode=info
canreinvite=yes
insecure=invite,port

extensions.conf

[globals]

[general]
static=yes
writeprotect=yes
autofallthrough=yes

[from-internal]

;internal extension 204 - rings for 30 seconds
exten => 204,1,Ringing()
exten => 204,n,Answer()
exten => 204,n,Dial(SIP/204,30,m)
exten => 204,n,Hangup()

;internal extension 8001 - rings for 30 seconds
exten => 8001,n,Ringing()
exten => 8001,n,Answer()
exten => 8001,n,Dial(SIP/8001,30,m)
exten => 8001,n,Hangup()

Sip Call mit anschließendem Disconnect

<--- SIP read from UDP:192.168.1.245:5060 --->
SIP/2.0 180 Ringing
Call-ID: [email protected]:5060
Contact: <sip:[email protected]:5060>
Content-Length: 202
Content-Type: application/sdp
CSeq: 102 INVITE
DependentInfo: 192.168.1.13
From: <sip:[email protected]>;tag=as0a221b1f
LeaveType: SD
MaxConnectingTime: 120
MaxLeaveWordTime: 90
MaxRingingTime: 30
ShortNumber: 204
To: <sip:[email protected]:5060>;tag=89362bc17bfa0086e62fd5a485bccf81
TransMode: SupportRTSP
User-Agent: Dahua UAC/3.0 VTH2421F_R V4.410.0.4
Via: SIP/2.0/UDP 192.168.1.219:5060;rport=5060;branch=z9hG4bK482e21be

v=0
o=0 0 0 IN IP4 192.168.1.245
s=Dahua VT 1.5
c=IN IP4 192.168.1.245
t=0 0
m=audio 20000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 20001 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
<------------->
--- (17 headers 10 lines) ---
sip_route_dump: route/path hop: <sip:[email protected]:5060>
Found RTP audio format 0
Found audio description format PCMU for ID 0
Found RTP video format 96
Found video description format H264 for ID 96
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.245:20000
Peer video RTP is at port 192.168.1.245:20001

<--- SIP read from UDP:192.168.1.245:5060 --->
SIP/2.0 200 OK
Call-ID: [email protected]:5060
Contact: <sip:[email protected]:5060>
Content-Length: 317
Content-Type: application/sdp
CSeq: 102 INVITE
From: <sip:[email protected]>;tag=as0a221b1f
To: <sip:[email protected]:5060>;tag=89362bc17bfa0086e62fd5a485bccf81
User-Agent: Dahua UAC/3.0 VTH2421F_R V4.410.0.4
Via: SIP/2.0/UDP 192.168.1.219:5060;rport=5060;branch=z9hG4bK482e21be

v=0
o=- 1591445533 2 IN IP4 192.168.1.245
s=Dahua VT 1.5
c=IN IP4 192.168.1.245
t=0 0
m=audio 20000 RTP/AVP 101 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 PCM/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 20001 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
a=recvonly
<------------->
--- (10 headers 15 lines) ---
Found RTP audio format 101
Found RTP audio format 0
Found RTP audio format 97
Found audio description format PCMU for ID 0
Found unknown media description format PCM for ID 97
Found audio description format telephone-event for ID 101
Found RTP video format 96
Found video description format H264 for ID 96
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.245:20000
Peer video RTP is at port 192.168.1.245:20001
sip_route_dump: route/path hop: <sip:[email protected]:5060>
Transmitting (NAT) to 192.168.1.245:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK13cb3a0d;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as0a221b1f
To: <sip:[email protected]:5060>;tag=89362bc17bfa0086e62fd5a485bccf81
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length: 0


---
Audio is at 11220
Video is at 192.168.1.245:20001
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Reliably Transmitting (NAT) to 192.168.1.13:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK19b569a8;rport
Max-Forwards: 70
From: <sip:[email protected]:5060>;tag=as38ff9773
To: <sip:[email protected]>;tag=1576777131
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 1624935437 1624935438 IN IP4 192.168.1.219
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 192.168.1.245
b=CT:384
t=0 0
m=audio 20000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
m=video 20001 RTP/AVP 96
a=rtpmap:96 H264/90000
a=sendrecv

---
Audio is at 19458
Video is at 192.168.1.13:20001
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding codec alaw to SDP
Reliably Transmitting (NAT) to 192.168.1.245:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK08a88694;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as0a221b1f
To: <sip:[email protected]:5060>;tag=89362bc17bfa0086e62fd5a485bccf81
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 293

v=0
o=root 139138240 139138241 IN IP4 192.168.1.219
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 192.168.1.13
b=CT:384
t=0 0
m=audio 20000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=maxptime:150
a=sendrecv
m=video 20001 RTP/AVP 96
a=rtpmap:96 H264/90000
a=sendrecv

---

<--- SIP read from UDP:192.168.1.13:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK19b569a8;rport=5060
From: <sip:[email protected]:5060>;tag=as38ff9773
To: <sip:[email protected]>;tag=1576777131
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Dahua UAC/3.0 VTO3221D V4.300.0.5
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.245:5060 --->
SIP/2.0 100 Trying
Call-ID: [email protected]:5060
Content-Length: 0
CSeq: 103 INVITE
From: <sip:[email protected]>;tag=as0a221b1f
To: <sip:[email protected]:5060>;tag=89362bc17bfa0086e62fd5a485bccf81
User-Agent: Dahua UAC/3.0 VTH2421F_R V4.410.0.4
Via: SIP/2.0/UDP 192.168.1.219:5060;rport=5060;branch=z9hG4bK08a88694

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

<--- SIP read from UDP:192.168.1.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK19b569a8;rport=5060
From: <sip:[email protected]:5060>;tag=as38ff9773
To: <sip:[email protected]>;tag=1576777131
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: Dahua UAC/3.0 VTO3221D V4.300.0.5
Content-Type: application/sdp
Content-Length: 204

v=0
o=0 0 0 IN IP4 192.168.1.13
s=Dahua VT 1.5
c=IN IP4 192.168.1.13
t=0 0
m=video 20001 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
m=audio 20000 RTP/AVP 0
a=rtpmap:0 PCMU/8000


<------------->
--- (10 headers 11 lines) ---
Transmitting (NAT) to 192.168.1.13:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK38eaaed1;rport
Max-Forwards: 70
From: <sip:[email protected]:5060>;tag=as38ff9773
To: <sip:[email protected]>;tag=1576777131
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.245:5060 --->
SIP/2.0 200 OK
Call-ID: [email protected]:5060
Contact: <sip:[email protected]:5060>
Content-Length: 308
Content-Type: application/sdp
CSeq: 103 INVITE
From: <sip:[email protected]>;tag=as0a221b1f
To: <sip:[email protected]:5060>;tag=89362bc17bfa0086e62fd5a485bccf81
User-Agent: Dahua UAC/3.0 VTH2421F_R V4.410.0.4
Via: SIP/2.0/UDP 192.168.1.219:5060;rport=5060;branch=z9hG4bK08a88694

v=0
o=0 0 0 IN IP4 192.168.1.245
s=Dahua VT 1.5
c=IN IP4 192.168.1.245
t=0 0
m=audio 20000 RTP/AVP 101 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 PCM/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 20001 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
a=recvonly
<------------->
--- (10 headers 15 lines) ---
Transmitting (NAT) to 192.168.1.245:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK4db02369;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as0a221b1f
To: <sip:[email protected]:5060>;tag=89362bc17bfa0086e62fd5a485bccf81
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.245:5060 --->
REGISTER sip:192.168.1.219 SIP/2.0
Authorization: Digest username="204", realm="asterisk", nonce="4ec7cdc3", uri="sip:192.168.1.219", response="69ae578920f63cd228ac79676a37c0e5", algorithm=MD5
Call-ID: 366afdc7b3e917daa6ebcb5b1ff4c37a
Contact: <sip:[email protected]:5060>
Content-Length: 0
CSeq: 73 REGISTER
Expires: 60
From: <sip:[email protected]:5060>;tag=51cfbce6af71c65e9af81d7a1b9ebddc
Max-Forwards: 70
PhoneState: 1
To: <sip:[email protected]:5060>
User-Agent: Dahua UAC/3.0 VTH2421F_R V4.410.0.4
Via: SIP/2.0/UDP 192.168.1.245:5060;rport;branch=z9hG4bKe75cd42afe19a99e56df7354afbf9309

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.245:5060 (NAT)
Sending to 192.168.1.245:5060 (NAT)

<--- Transmitting (NAT) to 192.168.1.245:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKe75cd42afe19a99e56df7354afbf9309;received=192.168.1.245;rport=5060
From: <sip:[email protected]:5060>;tag=51cfbce6af71c65e9af81d7a1b9ebddc
To: <sip:[email protected]:5060>;tag=as7c642c94
Call-ID: 366afdc7b3e917daa6ebcb5b1ff4c37a
CSeq: 73 REGISTER
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c4b4d4c"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '366afdc7b3e917daa6ebcb5b1ff4c37a' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.245:5060 --->
REGISTER sip:192.168.1.219 SIP/2.0
Authorization: Digest username="204", realm="asterisk", nonce="3c4b4d4c", uri="sip:192.168.1.219", response="d3592ab10b2ec672c1617bde875f7317", algorithm=MD5
Call-ID: 366afdc7b3e917daa6ebcb5b1ff4c37a
Contact: <sip:[email protected]:5060>
Content-Length: 0
CSeq: 74 REGISTER
Expires: 60
From: <sip:[email protected]:5060>;tag=51cfbce6af71c65e9af81d7a1b9ebddc
Max-Forwards: 70
PhoneState: 1
To: <sip:[email protected]:5060>
User-Agent: Dahua UAC/3.0 VTH2421F_R V4.410.0.4
Via: SIP/2.0/UDP 192.168.1.245:5060;rport;branch=z9hG4bKf4b2c32bdb3c773ed53e15e9286b4206

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.245:5060 (NAT)
Reliably Transmitting (NAT) to 192.168.1.245:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK0a111184;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as7d005d06
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Date: Sat, 06 Jun 2020 10:12:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 192.168.1.245:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKf4b2c32bdb3c773ed53e15e9286b4206;received=192.168.1.245;rport=5060
From: <sip:[email protected]:5060>;tag=51cfbce6af71c65e9af81d7a1b9ebddc
To: <sip:[email protected]:5060>;tag=as7c642c94
Call-ID: 366afdc7b3e917daa6ebcb5b1ff4c37a
CSeq: 74 REGISTER
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:[email protected]:5060>;expires=60
Date: Sat, 06 Jun 2020 10:12:26 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '366afdc7b3e917daa6ebcb5b1ff4c37a' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.245:5060 --->
SIP/2.0 200 OK
Call-ID: [email protected]:5060
Content-Length: 0
CSeq: 102 OPTIONS
From: "asterisk"<sip:[email protected]>;tag=as7d005d06
To: <sip:[email protected]:5060>;tag=5485dca56a46a998973bdc57cc9c8c72
User-Agent: Dahua UAC/3.0 VTH2421F_R V4.410.0.4
Via: SIP/2.0/UDP 192.168.1.219:5060;rport=5060;branch=z9hG4bK0a111184

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

 

[kuzco-ip]

Das Provlem hatte ich auch:

SIP Asterisk - VTO + 2 VTH

Hallo zusammen. Ich habe versucht einen SIP Server meinem Raspberry zu erstellen um meine VTO2000A und meine 2 VTH's einzubinden. Ich habe diese Anleitung genutzt (in welcher aber die Einrichtung der VTH leider nicht beschrieben ist)...

www.ip-phone-forum.de

 

[fluffymadness]

Danke für Link. Den Thread hatte ich bereits gelesen. Nur ich hab ein Video vor dem Abheben am VTH. Weil das Video was direkt nach dem Klingeln an der Glocke kommt wird über RTSP und nicht über SIP übertragen.
Sobald ich am VTH abhebe friert das Videobild am VTH ein (Standbild) weil er auf SIP Video umschalten will.
Und über SIP übertragt er mir unerklärlicherweise kein Video. Aber nur zum VTH. Andere Clients, Linphone etc bekommen vom VTO ein Video.

 

[kuzco-ip]

Ok, sorry, dann habe ich Dich missverstanden. Ich dachte, Du bekommst kein Video vor dem Abheben auf der VTH.

Ich für meinen Teil habe Asterisk auch mal getestet. Video auf VTH und Foto auf den Fritz-Fon funktionierte alles prima, aber:
wenn ich an einem Gerät den Anruf entgegengenommen habe, haben alle anderen Geräte weitergebimmelt - der Gruppenanruf wurde nicht beendet.
Ich habe für mich irgendwann dann entschlossen Asterisk als SIP wieder zu löschen und ohne die Fritz-Fon-Benachrichtigung zu leben.
Wenn Du hier eine Idee hast wie man das umsetzen kann, dann gerne. Ich denke Du hast mehr Ahnung von Asterisk als ich.

 

[Bluehorses]

For people who like to now how to setup the VTO3211D with RASPBX

1. I installed the VTO3211D and a VTH5441 on Raspbx with the following setting
2. Every device needs it’s own extension offcourse
3. Only use SIP protocol (pjsip is not working), you must use RFC2833, AVP And video enabled
4. If you want to use the door unlock function for the VTO extension use SIP INFO (dtmf-relay) and set the used key in the VTO
5. For transport use TCP and especially from the VTO with firmware 2019-10-19 V4.400.000000.3R use TLS (UDP gave no working setup)
6. Install some Linphone on your laptop or mobile and create a sip account by linphone for the push notification
7. Make a new CUSTOM extension in Raspbx for the push notification to Linphone and in the Advanced settings tab by dial set SIP/[email protected]
8. Change the behavior of the extensions where the VTO is calling for in such a way that after no answer the call will be redirected to the custom extension for a push notification Setup in stap 7

In the VTO I set a sFTP to my NAS for the saved security pictures.

Last edited: Today at 1:11 PM

 

[dahdah]

Seems I have the same issue like @Bluehorses (https://www.ip-phone-forum.de/threads/dahua-vto4202f-p-s2-asterisk-dahua-vth5422h-kein-video.314227/)
can you share your sip.conf and extensions.conf?

on point 3 what do you mean by AVP?

 

[dahdah]

@fluffymadness hast du das Problem gelöst?