>>> On 3/31/2009 at 4:38 PM, in message <[email protected]>, >>> Cuneyt M <[email protected]> wrote: > Hello everyone, > > We have existing setup of SipX 3.10.3 with AudioCodes gateways. > > However, we will be testing/integrating Bandwidth.com US Sip-Trunk (with > DID) with SipX. I understand that we can add a Sip Trunk just like > another gateway in Sipx(?) > In 4.0 the sipx system is able to communicate with ITSP's, there is already a preconfigured template for Bandwidth.com in the current 3.11x development version. In 3.10.x you will still need a sip trunking device, like an Ingate. > > The main use-case would be Philippines satellite-office with local SipX > server routing +1 numbers via the Bandwidth.com's SIP Trunk. > I assume we'll need one Sip Trunk per extension (if we are to assume > each extension will be sales guys calling their leads in US) - correct > me if im wrong. > Think of "trunks" as "lines". These are call paths. If you have a two-way trunk (inbound and outbound calling), then you can either receive or place a call on the trunk (line). If you have 20 people, and 5 are making calls and 5 are on calls that came in and 5 more are in the process of ringing, that's "15" simultaneous call paths. So for that kind of volume you need a minimum of 15 trunks and a 20-pack of DID numbers. Remember to spare some bandwidth for this, on tten simultaneous voice conversations 9where audio is playing of any sort, not ringing, like leaving or listening to voicemail, AA or speaking interactively with another person, then conservatively you need to budget roughly 800k for bandwidth in upload/download, and if you are sending it through a firewall you might look at prioritizing your traffic based on destination and port. > > We have no prior dealing with ITSP or SIP trunking as we used AudioCodes > GWs for local or close-proximity termination. > > Can you please share your thoughts and experience in setting up a Dial > Plan in SipX 3.10.3 where it'll route the call to the registered > ITSP(under Gateways) for numbers that start with +1 (US numbers). > I would think that the number would choose the gateway so to speak, seems straightforward. > > Also if you can share your experience on US-based Sip Trunk > providers(Bandwidth.com and others) for service reliability, features > and integration with SipX 3.10.3 i would very much appreciate. > We use them now with an Ingate, it's been very stable and pleasant, and the config for the Ingate is one of the least complicated I've seen. I do think you need a static IP address for bandwidth.com, and it makes registration issues a "non-issue". > Thanks in advance. > all the best! > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users >
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