[Freeswitch-users] WebRTC(VertoJS) to SIP Call Issue

Mon Aug 22 23:57:20 MSD 2016

If you have 2 different machines and trying to load balance between them,  say FS1 and FS2, then you need to do some additional steps to make sure FS2 can find users on FS1… usually the best way to do this is just send the calls to the machine the user is registered too…

There may be some more advanced HA ways to do this, but you start getting really complicated at that point and I would recommend contacting consulting at freeswitch.org <mailto:consulting at freeswitch.org>  to get one of the pro’s to help you with this sort of configuration

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Murugan Pandian
Sent: Monday, August 22, 2016 1:58 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] WebRTC(VertoJS) to SIP Call Issue

The same dialplan and dialstring  work fine in single FS,When i use two different FS then only i face this issue ,If my dialpaln or dialstring or incorrect it should not work single machine also right ,If i am not work 

On Tue, Aug 23, 2016 at 12:20 AM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

You either have your dialplan incorrect or your dialstring in the directory incorrect…. This does work as advertised if those 2 things are setup correctly.. see the example vanilla configs

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Murugan Pandian
Sent: Monday, August 22, 2016 1:48 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] WebRTC(VertoJS) to SIP Call Issue

Is there Any way i can achieve this 

On Mon, Aug 22, 2016 at 7:42 PM, Murugan Pandian <manpower13.cse at gmail.com <mailto:manpower13.cse at gmail.com> > wrote:

When i make call from FS-Verto to FS-SIP ,My FS-Verto syas Extension Not available 

On Mon, Aug 22, 2016 at 7:31 PM, Ítalo Rossi <italo at freeswitch.org <mailto:italo at freeswitch.org> > wrote:

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 at gmail.com <mailto:manpower13.cse at 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 at softov.com.br <mailto:fernando at 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 at gmail.com <mailto: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 <http://verto.yyyyy.com> 

2.FS-sip(SIP) in single Machine DNS EX:sip.yyyyy.com <http://sip.yyyyy.com> 

Client 1 (VertoJS) register successfully with FS-verto(verto.yyyyy.com <http://verto.yyyyy.com> )

Client 2 (Softfone)Register successfully with FS-sip(sip.yyyyy.com <http://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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-- 

Ítalo Rossi

italo at freeswitch.org <mailto:italo at freeswitch.org> 

_________________________________________________________________________
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