On Thu, Feb 26, 2009 at 03:21:54PM +1100, Amos Shapira wrote: > 2009/2/26 Ori Berger <linux...@orib.net>: > > Some information that may be useful if anyone is still interested: > > Thanks for the info. > > > > > - I can recommend grnvoip's termination service: They have good routes, good > > rates, competent technical support. They do not officially support IAX2 > > termination (only SIP and H323), but they will provide it if asked > > (supposedly; I'm using SIP termination). I heard great things about voipjet, > > but apparently they now actively require you to be a non-person entity > > (read, company) to join. > > I bought a $US50 credit with grnvoip and they sent me the exact > Asterisk configuration for my account when I asked. They also > initiated contact when they saw that I haven't used my credit within a > few days and offered support. That gave me an impression of good > service(TM). > > > > > - The cheap setup described by Arik is perfect for call _routing_ so long as > > the asterisk server is only there for routing, and can "step out" of the > > communication chain once call routing is finished. Otherwise, at least with > > a Xen setup on vpslink, the CPU slice is not regular enough to provide > > acceptable quality, even for things like a voicemail app. (Everything works, > > but sound is occasionally choppy). OpenVZ might be better; Lylix.net might > > be better; I only have experience with Xen, and it's NOT good enough. > > I setup a Xen guest with CentOS 5.2 and Asterisk 1.4.23(?) at VPSLink > and it's generally pretty accessible from where I am. > > Why do you think OpenVZ might be better? Does it have a known > advantage in Asterisk hosting? > > Googl'ing for "asterisk hosting provider" Lylix.net indeed comes up > near the top and seem to be asterisk-centric but their cheapest plan > of AsteriskNow is $35/month. No competition for the $8/month from > VPSLink. > > > > > - In order to enable Asterisk to step out of call routing (and network > > routing), the DID mapping protocol and the termination protocol must be the > > same -- either both should be IAX2 (when using VoipJet) or both should be > > SIP (when using grnvoip). Otherwise, asterisk will need to remain "on the > > line" to do protocol translation. > > > > - Asterisk rocks! It takes a little effort to configure, and looks weird at > > first (at least to my originally telephony-uninitiated self), but in most > > cases, there's a good reason for the way it needs to be configured. I think > > it's worthwhile to try to understand why Asterisk is built the way it is, > > rather than just look for an easy to configure GUI. > > I'm still struggling with Asterisk configuration. At some stage a > worker of mine got the sample echo test to work from my workplace but > that exhausted his Asterisk knowledge and I wasn't able to repeat that > test from home. > > I also tried to follow the Asterisk book > (http://www.the-asterisk-book.com/unstable/) and didn't manage to do > any dialing through even with the simplest, first configuration > example > (http://www.the-asterisk-book.com/unstable/minimale-telefonanlage.html#asterisk-konfigurieren) > > But I'm actually stuck at a more basic stage - I can't get incoming > audio on any of the software SIP clients I tried on my Ubuntu (8.10, > i386). I tried Twinkle (recommended here for its better logging), > Ekiga and Gizmo.
Let's start with something simpler: a call between the phone and Asterisk itself: an echo test, playback, voicemail extension, or whatever. -- Tzafrir Cohen | tzaf...@jabber.org | VIM is http://tzafrir.org.il | | a Mutt's tzaf...@cohens.org.il | | best ICQ# 16849754 | | friend _______________________________________________ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
