[Freeswitch-users] WebRTC(VertoJS) to SIP Call Issue
Luiz Fernando Softov
fernando at softov.com.br
Sun Aug 21 03:47:21 MSD 2016
To use in the same server, see
https://freeswitch.org/confluence/display/FREESWITCH/mod_verto
Permissions
and dial-string
<param name="dial-string"
value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})},${verto_contact
${dialed_user}@${dialed_domain}}"/>
To make calls between two servers, using presence, i don't know how.
I'm starting to use mod_verto, so I was trying to get user registration
from database
.
Then I
read the source of mod_verto, and figure out that verto doesn't manipule
database to save extensions registered.
So,
I think verto doesn't use database
and don't manipulate presence
.
Maybe it need some patch to be done, or already have a way.
2016-08-20 17:07 GMT-04:00 Murugan Pandian <manpower13.cse at gmail.com>:
> HI,
>
> I have two FS Accessing Same database(PSQL in CORE and Sofia odbc-dsn)
>
> 1.FS-verto(WebRTC) Vertojs in single Machine DNS EX:verto.yyyyy.com
> 2.FS-sip(SIP) in single Machine DNS EX:sip.yyyyy.com
>
>
> Client 1 (VertoJS) register successfully with FS-verto(verto.yyyyy.com)
> Client 2 (Softfone)Register successfully with FS-sip(sip.yyyyy.com)
>
> Issue:
>
> When i make call from Client 1(WebRTC) to Client 2 (SIP) its always say
> extension not available .
>
>
> But when i try this in single system it works,I stuck here Any one faced
> this kind of issue pls help me here
>
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