Hello, I am using Asterisk (16.2.1 on Debian 10 - 16.2.1~dfsg-1+deb10u1) with a FritzBox7590 (FW 7.12 lastest) and an IP-VideoDoorsystem (Dahua VT2000a FW:4.3; VTH1550CH FW:4.3).
As long as I have only 1 external Monitor (SIP/11032) in my dialplan everything is working fine. However, I would like to add my FritzFon C5 (11) into a grouplan - means the external Monitor and my FritzFon C5 should ring if I push my button (9901) on my IP-Videodoorsystem. So far so good.
BUT, if I cancel the call on my FritzFon C5 (11) or my VTH (SIP/11032), the IP-Videosystem and also the other device is still ringing. Only, if I cancel also the call on the other device, the VTO will also stop.
This means the hangup command does not hangup all device - just only the device where I have canncelled the call.
Please find below my sip.con, extensions.conf and the log during the call:
sip.conf:
[general]
language=de
bindport=5060
bindaddr=0.0.0.0
externrefresh=30
nat=force_rport,comedia
srvlookup=yes
transport=udp
;externip=[SipServerIP]
;localnet=10.40.0.1/255.255.0.0
externip=192.168.25.95
localnet=192.168.25.0/255.255.255.0
directmedia=yes
videosupport=yes
;progressinband=yes
;internal_timing=yes
;silencesuppression=no
; register => [UsernameInFritzBox]:[PasswordInFritzBox]@[IPofFritzBox]/[UsernameInFritzBox]
register => vto2000a35:1122334455@192.168.25.1/vto2000a35
; VTO2000A - .35 - RoomNr. 8001
[8001]
host=dynamic
defaultuser=VTO2000A
type=friend
;secret=[PasswordForVTO]
secret=6677889900
context=sip-out
canreinvite=yes
qualify=yes
disallow=all
allow=ulaw
allow=h264
dtmfmode=info
callerid=VTO2000A-35 <8001>
; VTH - SB - .32
[11032]
host=dynamic
defaultuser=VTO2000A
type=friend
;secret=[PasswordForVTO]
secret=6677889900
context=sip-out
canreinvite=yes
qualify=yes
disallow=all
allow=ulaw
dtmfmode=info
videosupport=yes
;progressinband=yes
;internal_timing=yes
;silencesuppression=no
;directmedia=yes
;directrtpsetup=yes
;prematuremedia=no ;this does the exact opposite of what everyone assumes it does
;progressinband=no
callerid=VTH Spitzboden <11032>
; sorgt dafuer dass die internen FritzFon angerufen werden
[videodoorgateway]
context=sip-in
type=friend
insecure=invite
nat=force_rport,comedia instead
;defaultuser=[UsernameInFritzBox]
;fromuser=[UsernameInFritzBox]
defaultuser=vto2000a35
fromuser=vto2000a35
fromdomain=fritz.box
;secret=[PasswordInFritzBox]
;host=[IPofFritzBox]
secret=1122334455
host=192.168.25.1
dtmfmode=rfc2833
disallow=all
allow=ulaw
extensions.conf:
[general]
static=yes
writeprotect=no
[sip-out]
; NoOp notwendig, da die VTH sonst erst Video nach dem Abheben zeigt !
; necessary, otherwise the VTH will only show the video after accepting the call
exten => 9901,1,NoOp(doorbell ring group)
;exten => 9901,1,Set(CALLERID(num)=9901)
exten => 9901,n,Ringing()
exten => 9901,n,Answer()
; exten => 9901,n,Set(CALLER=${CALLERID(num)})
exten => 9901,n,Dial(SIP/11@videodoorgateway&SIP/11032,30,m)
exten => 9901,n,Hangup()
exten => 8001,1,Dial(SIP/8001) ; VTO-35
exten => 11031,1,Dial(SIP/11031) ; VTH-31
exten => 11032,1,Dial(SIP/11032) ; VTH-32
exten => 11033,1,Dial(SIP/11033) ; VTH-33
[default]
include => sip-out
LOG
asterisk*CLI>
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
> 0x563510319230 -- Strict RTP learning after remote address set to: 192.168.25.35:20000
> 0x563510298bb0 -- Strict RTP learning after remote address set to: 192.168.25.35:20001
-- Executing [9901@sip-out:1] NoOp("SIP/8001-000000c0", "doorbell ring group") in new stack
-- Executing [9901@sip-out:2] Ringing("SIP/8001-000000c0", "") in new stack
-- Executing [9901@sip-out:3] Answer("SIP/8001-000000c0", "") in new stack
> 0x563510319230 -- Strict RTP switching to RTP target address 192.168.25.35:20000 as source
-- Executing [9901@sip-out:4] Dial("SIP/8001-000000c0", "SIP/11@videodoorgateway&SIP/11032,30,m") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Called SIP/11@videodoorgateway
-- Called SIP/11032
-- Started music on hold, class 'default', on channel 'SIP/8001-000000c0'
-- SIP/11032-000000c2 connected line has changed. Saving it until answer for SIP/8001-000000c0
> 0x7ff5c8008300 -- Strict RTP learning after remote address set to: 192.168.25.1:7082
-- SIP/videodoorgateway-000000c1 is making progress passing it to SIP/8001-000000c0
> 0x7ff5c8008300 -- Strict RTP switching to RTP target address 192.168.25.1:7082 as source
> 0x563510298bb0 -- Strict RTP switching to RTP target address 192.168.25.35:20001 as source
> 0x7ff5c800b630 -- Strict RTP learning after remote address set to: 192.168.25.32:20000
> 0x7ff5c800ed80 -- Strict RTP learning after remote address set to: 192.168.25.32:20001
-- SIP/11032-000000c2 is ringing
-- SIP/11032-000000c2 is making progress passing it to SIP/8001-000000c0
> 0x7ff5c800ed80 -- Strict RTP switching to RTP target address 192.168.25.32:20001 as source
-- Got SIP response 486 "Busy Here" back from 192.168.25.1:5060
-- SIP/videodoorgateway-000000c1 is busy
-- Got SIP response 486 "Busy Here" back from 192.168.25.32:5060
-- SIP/11032-000000c2 is busy
== Everyone is busy/congested at this time (2:2/0/0)
-- Stopped music on hold on SIP/8001-000000c0
-- Executing [9901@sip-out:5] Hangup("SIP/8001-000000c0", "") in new stack
== Spawn extension (sip-out, 9901, 5) exited non-zero on 'SIP/8001-000000c0'
asterisk*CLI>
Hopefully someone knows how to cancel a groupcall which includes DECT and SIP devices.