Hikvision DS-KV8213-WME1 ONVIF not working

NoFate said:
1) why is that canreinite important?
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I don't remember full details, sorry, I believe, on linphone app there was no video at all without this option.
The reason could be that my asterisk server is sitting behind strong NAT without white IP address.
Probably you can find some explanation here Asterisk sip.conf canreinvite option
2) in your dialplan, if you call 101 , do 101 and 102 and linphone user all ring simultanously? i thougth that that didnt work, that only 1 extension was ringing , if 101 is not avaible , then goto 102
So do they all ring at the same time? and with early vidoe?
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Yes, all online phones ring simultaneously. If nobody pickup the call within 30 secs or all ## are offline, asterisk start calling next linphone #.
For the early video I have to double check. But it definitely come to my phone (#102) and yyyyyyy@linphone.org, if I'm away from home.
3) i also see you call externally a linhone user based on sip.linphone.org, how do you that ? i also call linhome users; i created an outbound trunk that registers on linhome flexisip, with that trunk i do an outbound call
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Honestly speaking, I didn't make any sip.linphone.org specific setup. It just works out of the box.
 
ok, i try that linphone thing, i though you needed a trunk for that ...
i have loaded pjsip, gonna try to recompile with chan_sip again, seems that canreinvite is indeed important for direct_media
seems canreinvite=no is the same as direct_media=no for pjsip, but after googling, seems a lot of users have issues with pjsip and direct_media

why s your dialplan like:
exten => 101/100,1 ....
why not just:
exten => 101,1
whats the difference?
 
NoFate said:
why s your dialplan like:
exten => 101/100,1 ....
why not just:
exten => 101,1
whats the difference?
Click to expand...
The difference is the chain processing, when the call is coming from the Intercom (SIP #100), which is not required for the calls to SIP #101 from other SIP ##.
 
strange, that you can call linhome/linhome users directly

if i do like;

Code: 
exten => 100,1,NoOp()
 same => n,Dial(PJSIP/xxxxo@sip.linhome.org)

i get:

Code: 
[Nov  9 15:35:27] NOTICE[418][C-00000001]: app_dial.c:2707 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)

i need to setup a trunk first that registers on linhome/linphone, and then i do an outbound call using that trunk
 
tested verry quickly with pjsip and direct_media= off/on
as soon i do :

Code: 
exten => 6001,1,NoOp()
 same => n,Progress()
 same => n,Dial(PJSIP/6001&PJSIP/6002)

6001 and 6002 but only video after pickup
if i just call 6001 OR 6002 , i do see ealy media

(just testing with softphones for now)

time to test chan_sip ... need to install/compile chan_sip, since i run asterisk as a docker
something for next time

fingers crossed
 
SIP doesnt work, i need PJSIP, i dont have chan_sip loaded on my asterisk

yes, 6001 and 6002 do ring together, but without early video
 
PS The configuration above certainly contains only SIP ##100/101 related processing.

not sure what that means? is ie also related to early_video , dont think so right?
 
i runk Asterisk as addon on Home Assistant :-), so need to change dockerfile to load chan_sip again, it was removed from it
 
yeah, maybe, if i change ports

but doesnt mather, i have an older dockerfile with chan_sip, gonna load a a new docker, and try later :)
hopefully that canreinvite was the cullprit, thats the only thing left to test for me
 
Can you show me some outbound trunk configuration, to get idea what to look for. May be it was set already when I install asterisk in Ubuntu.
 
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sure, its all based on pjsip, more code that chan_sip
then in exensions i call another user (yyyy) using this registered user below (xxx)

Code: 
 same => n,Dial(PJSIP/yyyy@trunk-linhome)


Code: 
; ################# TRUNK Linhome #####################

[trunk-linhome-auth]
type=auth
auth_type=userpass
password=Password01-
username=xxxx

[trunk-linhome-aor]
type=aor
contact=sip:sip.linhome.org
;qualify_frequency=30

[trunk-linhome-registration]
type=registration
outbound_auth=trunk-linhome-auth
server_uri=sip:sip.linhome.org
client_uri=sip:xxxx@sip.linhome.org
contact_user=xxxx.linhome
retry_interval=30
 
[trunk-linhome]
type=endpoint
context=default
disallow=all
allow=ulaw,alaw
allow=h264,vp8
outbound_auth=trunk-linhome-auth
aors=trunk-linhome-aor
from_domain=sip.linhome.org
direct_media=yes
 
[trunk-linhome-identify]
type=identify
endpoint=trunk-linhome
match=sip.linhome.org
 
Well, pjsip.conf here (without any changes, untouched after installations) contains some trunk related settings, but all of them are commented. No any in the sip.conf.
And as you can see from my settings, chan_sip makes external calls directly to linphone ident ;).
 
Yeah, gonna try later again, I have found a slightly different syntax:

community.asterisk.org

Resolving issue

I have two records in dialplan: exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall@sip.linphone.org) exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall@iptel.org) Calling testA works fine, trying to call testB gives: -- Executing [testB@context:1] Dial("PJSIP/client-00000002"...
community.asterisk.org community.asterisk.org

Can you maybe post your full config for 100 and 101? Gonna copy past later when I have chan_sip running... Maybe also your transport section?
You only have the canreinvite=no on the 100 extension?
 
NoFate said:
Can you maybe post your full config for 100 and 101? Gonna copy past later when I have chan_sip running... Maybe also your transport section?
Click to expand...
I updated code under spoiler: add missing section of the extensions.conf please check.
transport= is commented at working sip.conf. I can see some previous sip.conf.bak, which is not in use, with transport=udp, so...
NoFate said:
You only have the canreinvite=no on the 100 extension?
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Yes, only in this [100] section. Other clients seems don't have problem with video without canreinvite=no.
 
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