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

Tue Aug 23 00:08:26 MSD 2016

Ken Thanks for your quick reply,We are using load balancing with two FS But
both FS share same Postgres (Mean  sofia odbc-dsn) meaning every user
present available in table if i am not wrong ,then why i need to register
two FS and we are using mod_xml_curl for control freeswitch.

We need to enable presence also in ODBC ?,Because we already enable sofia
odbc-dsn enabled

Thanks

On Tue, Aug 23, 2016 at 1:27 AM, Ken Rice <krice at freeswitch.org> wrote:

> If you have 2 different machines and trying to load balance between them,
> say FS1 and FS2, then you need to do some additional steps to make sure FS2
> can find users on FS1… usually the best way to do this is just send the
> calls to the machine the user is registered too…
>
>
>
> There may be some more advanced HA ways to do this, but you start getting
> really complicated at that point and I would recommend contacting
> consulting at freeswitch.org to get one of the pro’s to help you with this
> sort of configuration
>
>
>
> *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 1:58 PM
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] WebRTC(VertoJS) to SIP Call Issue
>
>
>
> 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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>
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>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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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
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> http://www.cluecon.com
>
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>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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>
> _________________________________________________________________________
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>
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