[Vtigercrm-developers] A new approach to telephony integration

[Sreenivas Kanumuru]

Alan,

Thanks for sharing this. We are looking at Asterisk
-Java<http://www.asterisk-java.org/development/> to
improve the Asterisk integration in the open source product. But, OpenSIPs
looks better from what you shared, and the fact that it is not just
restricted to Asterisk is a plus. We will keep you posted if we make some
progress on OpenSIPs.

thanks,
Sreenivas

--
Sreenivas Kanumuru
vtiger Team

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On Fri, Sep 27, 2013 at 1:31 PM, Alan Bell <alan.bell at libertus.co.uk> wrote:

> Hi all,
>
> We have been using the vtiger click to call integration for a few
> customers and I have customers wanting inbound screen pop to bring up
> contact details based on callerID information. The current setup is rather
> dependent on an asterisk server with an available API or AMI interface and
> I want to do something for the situation where there is a hosted SIP
> provider where we have no access to it at all, and fairly dumb SIP handsets.
>
> I have been investigating OpenSIPs, which is a SIP proxy, it just handles
> signalling, not the voice calls themselves, so it is does not require a lot
> of bandwidth or CPU. I have this set up so that it works as a pure proxy,
> the handsets talk to it as if it was the upstream PBX, it sends all their
> signalling upstream, all the return signals are passed back to the handset.
> This basically means it sits in the middle between handsets and exchange
> and it can see all the calls being initiated. It can't see the VOIP traffic
> itself, that goes point to point. It can fire events when a call happens,
> which can include XMLRPC calls, so it can poke the vtiger server with a
> from address and the dialled number. Having got that into the vtiger server
> we can log the call and we could use a websockets connection to the end
> user browser to cause an immediate popup with the contact details on
> incomming calls. We can also do click-to-call in a similar SIP friendly
> way, by using the PHP-SIP library to send an INVITE to the user extension,
> then a REFER to make the extension dial the outbound number, then a BYE to
> hand off the call.
>
> Basically this means we can do inbound and outbound call handling with
> zero dependence on support from the SIP exchange - and it doesn't have to
> be asterisk or open source, any proprietary SIP platform will do. The
> handsets that want to integrate with vtiger get reconfigured to point at
> the vtiger server rather than directly at their exchange, other handsets
> are not affected. This means in a big organisation we don't need to
> negotiate for direct admin level access to the exchange, we just
> reconfigure a few handsets in the sales and marketing department.
>
> I have most of the bits in place to prove I can do this, but there is a
> bit of a bug in openSIPs https://github.com/OpenSIPS/**opensips/issues/87<https://github.com/OpenSIPS/opensips/issues/87>which means it doesn't send valid XMLRPC at the moment, plus I have all the
> vtiger integration and websockets work left to do.
>
> If anyone has any feedback on this approach, or wants to collaborate on it
> then do get in touch.
>
> Alan Bell
>
>
> The openSIPs website: http://opensips.org/
> How the click to call bit works http://level7systems.co.uk/en/**
> blog/Click+to+Call+with+PHP-**SIP<http://level7systems.co.uk/en/blog/Click+to+Call+with+PHP-SIP>
>
>
> --
> Libertus Solutions
> http://libertus.co.uk
>
> ______________________________**_________________
> http://www.vtiger.com/
>
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