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

Murugan Pandian manpower13.cse at gmail.com
Mon Aug 22 23:22:57 MSD 2016


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 at gmail.com>
wrote:

> 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> 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] *On Behalf Of *Murugan
>> Pandian
>> *Sent:* Monday, August 22, 2016 1:48 PM
>> *To:* FreeSWITCH Users Help <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> 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>
>> 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> 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> 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>:
>>
>> 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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>>
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>>
>>
>> --
>>
>> Ítalo Rossi
>>
>> italo at freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
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>> http://www.cluecon.com
>>
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>>
>>
>>
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>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
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>
>
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