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

[Murugan Pandian]

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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>>>> http://www.freeswitchsolutions.com
>>>>
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>>>
>>>
>>> ____________________________________________________________
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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
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
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>
>
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