What does sofia_contact says for your sip ext on FS-verto box?

On Sun, Aug 21, 2016 at 2:36 AM, Murugan Pandian <manpower13.cse@gmail.com> wrote:
Hi Luiz,


 Hi don't have Any issue in single server when we Go with Two server then only i face this issue,Maybe freeswitch team will help here

On Sun, Aug 21, 2016 at 5:17 AM, Luiz Fernando Softov <fernando@softov.com.br> wrote:
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@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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