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

[Murugan Pandian]

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
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
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>>>> http://www.cluecon.com
>>>>
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>>>> switch-users
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>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
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>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
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>>>
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>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
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>>
>
>
>
> --
> Ítalo Rossi
> italo at freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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