The dialstring in the directory tells the user/ virtual endpoint the different ways to lookup where a user might be… it should look something like this in your directory
<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})}"/>
Not it does lookups for sofia and verto users… this data is then used for bridge to complete the call when you do something like
<action application=”bridge” data=”user@domain.com” /> where user@domain.com is the user from the user directory you are trying to call. This will then locate that user if they are connected via SIP or Verto….
If you are trying to do something else in the dialplan with your bridge app data it will probably not work
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Murugan Pandian
Sent: Monday, August 22, 2016 2:23 PM
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] WebRTC(VertoJS) to SIP Call Issue
We need to use different dialplan or dialstring for multiple FS(sofia odbc_cdn) ?
On Tue, Aug 23, 2016 at 12:28 AM, Murugan Pandian <manpower13.cse@gmail.com> wrote: