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@
lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Murugan Pandian
Sent: Monday, August 22, 2016 1:48 PM
To: FreeSWITCH Users Help <freeswitch-users@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@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@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@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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