Asterisk as an Aplication server /handing calls to gateway

first of all: I am new and please forgive my english, I am from germany:
Here´s the sceanrio / question:
Currently we use an IP-Centrex-platform from broadsoft. It´s ok, but we miss certain features. also, since we don´t run our own servers, we cannot control all features, licenses etc. by oursevles but depend on our partner.

we now think about ways to increase overall control about features and services, but do not want to install our own media-gateways.

is it possible or has anyone ever tried and succeeded in using asterisk as an application and media server which hands off all outgoing calls to media-gateways of another provider / partner ?

thanks in advance

sure this is done alot, by anybody that uses Asterisk with an ITSP (voip provider). This will work fine just make sure you have enough bandwidth to handle it and you have qos controls set up…

that basically the VoIP Provider just makes sip-ID´s available. those are configured on the asterisk-machine, which then enables all kinds of features, and services.
does that mean it would be possible to use features that are not supported by the original VoIP Provider ?
how does that scale ? could it be possible, given the fact that there is enough bandwith, to srevice a few thousand sip-id´s ?

  1. yes it means you can offer your users features that are not supplied by your ITSP. With Asterisk, all you really need the ITSP for is phone service, origination/termination. Any features you need to offer your users can be done wtih * via a number of means.
    Not sure what you mean by SIP ID, i assume you mean sip account. I wouldn’t recommend having Asterisk handle a few thousand SIP registrations, although with a decent machine it should be able to hack it. Generally if you have that much SIP traffic you put OpenSER in front of * and it deals with the SIP stuff, only talking to * when there is an actual call.
    And yes if you have the bandwidth and machine power, it will scale on up quite nicely.

However if you are providing your own services there is no real reason why you need thousands of SIP accounts. Just move your DIDs (phone numbers) to any number of ITSP wholesale providers, they will give you one set of SIP cridentials that can handle as many calls in or out as you throw at it (SIP trunking). It will also send you the incoming calls from your DIDs and asterisk can nicely sort them out. Thus you dont need thousands of little SIP accounts, just one for the trunk in/out, and more for your users (which may end up being thousands, see the bit about SER).

hoep taht helps

what will be the config if calls will be coming from another type of sip server and forward it to another SIP server? Basically forward the call to to another gw or ip.

thats easy, then you only need two SIP entries, one for each server. things like DID, callerid, etc can be preserved through this.

Both originating and terminating should be present in the sip.conf.

How about the exntension.conf? How will i forward the call from the originating provider to my terminating provider?

if you want asterisk to do nothing than blindly pass calls thru

exten => _X.,1,Dial(SIP/provider2/${EXTEN})

exten => _X.,1,Dial(SIP/provider1/${EXTEN})

and of course you can add other stuff to spice it up, call processing, voicemail, recording, whatever