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[Freeswitch-users] SIP Proxy with direct media path


 
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gromovd at gmail.com
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PostPosted: Sun Nov 01, 2009 12:21 am    Post subject: [Freeswitch-users] SIP Proxy with direct media path Reply with quote

Hi!

I was a long time user of MySipSwitch service (www.mysipswitch.com) but it looks like they are planning to close their service. I can't use the replacement they have and would actually prefer some locally installed solution.

Question - is it possible to use FreeSwitch exactly like MySipSwitch - just SIP Proxy with very flexible dial plan?
With MySipSwitch it is possible to:
- register service with multiple SIP providers as client
- register your ATA with MySipSwitch
- create smart dial plan (in Ruby) to route calls based on prefixes and/or call source and some other logic
- accept and route calls from all registered providers

MySipSwitch does not have any media capabilities - RTP is always established directly between call parties (no re-Invites either).

If this is possible - is there any beginner guide for this?

Also - I know that FreeSwitch could run in Windows as well as in Linux - would it be possible to use the above functionality in both OSes?

Thanks,
Dmitry
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DG
NJ
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jason at jasonjgw.net
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PostPosted: Sun Nov 01, 2009 12:30 am    Post subject: [Freeswitch-users] SIP Proxy with direct media path Reply with quote

Quote:
With MySipSwitch it is possible to:
- register service with multiple SIP providers as client

This is easy once you understand the FreeSWITCH XML configuration.
Quote:
- register your ATA with MySipSwitch

That's really a special case of the above.
Quote:
- create smart dial plan (in Ruby) to route calls based on prefixes and/or
call source and some other logic

FreeSWITCH can do better than that: there are lots of variables which can be
matched by Perl-compatible regular expressions, including the destination
number.
Quote:
- accept and route calls from all registered providers

That's handled in the same way as the previous item.
Quote:

MySipSwitch does not have any media capabilities - RTP is always established
directly between call parties (no re-Invites either).

FreeSWITCH gives you a choice: you can use proxy media or bypass media modes
if you don't want it to handle the media.
Quote:

If this is possible - is there any beginner guide for this?

http://wiki.freeswitch.org/ is the best there is.

It is being improved by the community over time.

You can also take advantage of the IRC channel, the FreeSWITCH conference and
of course the mailing list.


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gromovd at gmail.com
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PostPosted: Sun Nov 01, 2009 10:31 am    Post subject: [Freeswitch-users] SIP Proxy with direct media path Reply with quote

Hi!

On Sun, Nov 1, 2009 at 01:21, Jason White <jason@jasonjgw.net (jason@jasonjgw.net)> wrote:
Quote:

http://wiki.freeswitch.org/ is the best there is.

It is being improved by the community over time.

You can also take advantage of the IRC channel, the FreeSWITCH conference and
of course the mailing list.



Thank you very much for the reply... I had some experience with FreeSwitch about a year ago, but did not have more time to check all the features. And there wasn't much documentation available then.
I'll give it another try this time.

Thanks,
Dmitry

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DG
NJ
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