tjc at wintrmute.net
Wed Oct 18 18:07:04 BST 2006
On Wed, Oct 18, 2006 at 05:23:03PM +0100, Dirk Koopman wrote:
> On Wed, 2006-10-18 at 15:57 +0100, Toby Corkindale wrote:
> > Hey,
> > Is there any service around that will route real, live telephone numbers
> > through to a server's IP, via internet, using some kind of open-standard
> > protocol? (Eg. SIP)
> Are we talking existing telephone numbers or a new block?
> > Not just for a single phone number, but a whole block.
> > ie. So that one can run an office PBX without needing to buy an ISDN or SS7
> > card, and all the faffing about with that kind of thing.
> And are you aware of the issues?
I worked with (australian) telecoms for 4 years, so I know many issues around
ISDN/SS7, but not so much with the VOIP stuff.
> * You need a line that does not have any other services (eg web, mail,
> ftp) etc. Otherwise someone sending you a large email (for instance)
> will completely wipe out any other incoming RTP (whilst letting them
> still here you - most frustrating).
This is manageable with QoS, which I'm quite familar with.
> * It also helps if people in your organisation don't use that line for
> downloading huge volumes of stuff (otherwise the opposite problem occurs
> - it will clobber outgoing RTP). Having the correct router would allow
> you to use QoS to fix this.
> * You need an ISP that understands the issues and is happy to let you do
> it in the first place. More and more are doing their own
> "triple"/"quadruple" play thing and are therefore reluctant / unable /
> don't understand how to do useful things like QoS at their end (which
> would mitigate the first issue).
Good point; we're going to be switching ISP providers shortly anyway, so I'll
make sure they support QoS.
> * Using asterisk works well, except that it needs to be configured
> correctly, by someone clueful. A wrong entry may stop the whole thing
> working and the config files (particularly for a large organisation) are
> long and easy to get wrong. There are solutions to this of varying
> efficacy, but I still do it the "hard" way.
Yeah, but that goes for most things.
> * If you go asterisk and RTP then go to someone like Voiptalk.org /
> Magrathea (same people really) and use IAX. It's more efficient. You
> only really need one number, because you can multiplex several
> conversations down the same channel.
Cheers - that IAX service looks exactly like the sort of thing I'm after..
> Finally: don't dismiss using one of Digium's E1 cards and getting a real
> telephone line in. That asterisk config file is actually one of the
> shortest and easiest to deal with. It also gives you guaranteed
> bandwidth. Dealing with all the "issues" of WAN based VoIP will cost you
> quite a bit to do properly and get the quality that I suspect you will
> need. A real E1 line and a line card will be better quality, less hassle
> and cheaper (at least for incoming calls). You can still use VoIP
> to/from the desktops and you can mix and match the real telephone lines
> with outgoing VoIP calls.
I'm definately more familiar with that end of things; unfortunately it requires
higher setup costs, and I'm essentially working for a pre-funding startup here
CHeers for the info!
Turning and turning in the widening gyre/The falcon cannot hear the falconer;
Things fall apart, the centre cannot hold/Mere anarchy is loosed upon the world
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