[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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