Re: Setting up a PBX for Israel-US communication
Hello, It is the most compressed out of all the audio codecs As a matter of fact and as far as I know, the G.729 codec and G.729A codec are more compressed than gsm. (It could be that it is not supported by twinkle or other clients, I don't know ) See: http://en.wikipedia.org/wiki/G.729 Regards, Rami Rosen ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Yes I'm very interested though I haven't got around to get my asterisk working for me, and family say that for now they are happy with skype so I don't need the Isareli number. Thanks for the update. On 3/19/09, Ori Berger linux...@orib.net wrote: Amos Shapira wrote: BTW - About Israeli DID's - I just received an ad from Gizmo5 (I probably registered with them a while ago) advertising their new free SIP-to-Skype gateway. So I went to their web site and found that they offer a callin number for $12/3-months or $35/12-months, which is very close to the lost 077 numbers for $3/month with DIDWW. These are 03 numbers (http://gizmo5.com/pc/network/callin-numbers/). Anyone still interested in this thread -- it seems DIDWW has acquired another number block from Hot. Numbers still available at $3/month (+$3/setup). ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Amos Shapira wrote: BTW - About Israeli DID's - I just received an ad from Gizmo5 (I probably registered with them a while ago) advertising their new free SIP-to-Skype gateway. So I went to their web site and found that they offer a callin number for $12/3-months or $35/12-months, which is very close to the lost 077 numbers for $3/month with DIDWW. These are 03 numbers (http://gizmo5.com/pc/network/callin-numbers/). Anyone still interested in this thread -- it seems DIDWW has acquired another number block from Hot. Numbers still available at $3/month (+$3/setup). ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Fri, Feb 27, 2009 at 12:47:15AM +0200, sara fink wrote: In twinkle, try to use audio codec gsm. It is the most compressed out of all the audio codecs. I also have a ekiga account and want to try it with twinkle. If you have problems of bandwidth, then gsm should help. In most cases 128kbps upload is very problematic. If speex is supported, use it. Should give better compression and quality than gsm. -- 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
Re: Setting up a PBX for Israel-US communication
2009/2/26 Tzafrir Cohen tzaf...@cohens.org.il On Thu, Feb 26, 2009 at 03:21:54PM +1100, Amos Shapira wrote: Why do you think OpenVZ might be better? Does it have a known advantage in Asterisk hosting? Because your provider may be able to provide you basically the same resources for less money. So if OpenVZ is good enough for you: fine. You mean in case that the hosting provider provides the necessary kernel modules in OpenVZ? Is there any technical obstacle preventing an interested hosting provider from offering the required modules in Xen guest kernels? As for the cost - as far as I'm aware OpenVZ costs the same as Xen hosting at VPSLink and my personal experience with openvz/virtuozo hosting was that it's a bit limited near the edges. Oh, and as I mentioned non-dahdi (non-zaptel) timing for Asterisk, see the following for more information: http://lists.digium.com/pipermail/asterisk-dev/2009-February/036659.html Thanks. But before I can worry about all this - I still have to get audio back to my SIP clients at home, which I don't manage to do. Cheers, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Thu, Feb 26, 2009 at 08:32:51PM +1100, Amos Shapira wrote: 2009/2/26 Tzafrir Cohen tzaf...@cohens.org.il On Thu, Feb 26, 2009 at 03:21:54PM +1100, Amos Shapira wrote: Why do you think OpenVZ might be better? Does it have a known advantage in Asterisk hosting? Because your provider may be able to provide you basically the same resources for less money. So if OpenVZ is good enough for you: fine. You mean in case that the hosting provider provides the necessary kernel modules in OpenVZ? Kernel modules? What kernel modules? :-) (See the second part. Well, maybe this is more for future installations than for now) Is there any technical obstacle preventing an interested hosting provider from offering the required modules in Xen guest kernels? As for the cost - as far as I'm aware OpenVZ costs the same as Xen hosting at VPSLink and my personal experience with openvz/virtuozo hosting was that it's a bit limited near the edges. OpenVZ consumes less resources. Hence you can use more instances per server. Hence it costs less. VPSLink seem to price them the same. Most others don't from my previous searches. -- 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
Re: Setting up a PBX for Israel-US communication
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
Re: Setting up a PBX for Israel-US communication
2009/2/26 Tzafrir Cohen tzaf...@cohens.org.il: 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. That's exactly what I tried. I don't know how to configure asterisk to have a simple echo test working. Just a few minutes ago I called ekiga.net from Twinkle and for the first time I heard the instructions played by the echo test (I couldn't get even Ekiga itself to do that). But they are VERY chopy. Probably wrong audio device but only ALSA/default is accepted by Twinkle (not pulse) option. Do you have a simple asterisk echo test setup I can use? And how to configure Twinkle to connect to such a server? --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Thu, Feb 26, 2009 at 09:45:28PM +1100, Amos Shapira wrote: 2009/2/26 Tzafrir Cohen tzaf...@cohens.org.il: 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. That's exactly what I tried. I don't know how to configure asterisk to have a simple echo test working. The sample configuration has a context called [demo] . Send your phone to it. It's a sample IVR with prompts and such. It has an echo test extension as '500'. -- 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
Re: Setting up a PBX for Israel-US communication
Amos Shapira wrote: 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. They do have a non-managed $15/month plan (that still includes ztdummy and friends), comparable to what you get from vpslink at $15 -- however, they do not have a $8/month plan like vpslink. 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. I haven't done it myself, but apparently there are recommendations to open UDP ports 1-2 to your PC. I wouldn't. I use an ADSL2+ with D-Link DSL-G604T modem/router. I also have a Sipura ATA-3000 connected to my ISP's VoIP (SIP) service with no problem (and no port forwarding required in the modem). Is it possible the Sipura uses UPNP or something to punch holes in the firewall itself? Is it possible that they both use the same UDP port (e.g. 5000 or 5004), and for whatever reason the sipura gets the packets sent to your softphone? ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Tzafrir Cohen wrote: You mean in case that the hosting provider provides the necessary kernel modules in OpenVZ? Kernel modules? What kernel modules? :-) (See the second part. Well, maybe this is more for future installations than for now) Well, following Tzafrir's lead, I've just installed a 1.6.1-rc1 on my vpslink server with a res_timing_pthread time source, and the sound is slightly less clear but had no choppy sounds in the 5 minutes I played with it. I assume this has more to do with the time source than the 1.6 upgrade, but I don't know that. It's possible that a ztdummy driver on openvz provides a good enough time source that makes it work better than Xen+pthread -- if that's the case, lylix.net would be superior. A friend of mine is trying them now, we'll see how that works. As far as vpslink is concerned, I've been unsuccessful in getting a ztdummy to work. It's a Xen setup, but you can only use a very specific centos kernel, no matter what system you are running -- just getting the ztdummy to compile with the required gcc-4.1 for that kernel is a lot of effort. And when the module didn't work, I gave up. OpenVZ consumes less resources. Hence you can use more instances per server. Hence it costs less. VPSLink seem to price them the same. Most others don't from my previous searches. Tzafrir, do you know of any OpenVZ provider that charges something comparable to the $8/month vpslink plan? ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
2009/2/26 Ori Berger linux...@orib.net: OpenVZ consumes less resources. Hence you can use more instances per server. Hence it costs less. VPSLink seem to price them the same. Most others don't from my previous searches. Tzafrir, do you know of any OpenVZ provider that charges something comparable to the $8/month vpslink plan? I know someone who uses BlueHost to host his blog for years (started at $6/month, maybe new users will pay a little more) and is very satisfied. I'm NOT sure this is an OpenVZ product but it's something similar (jailed environments sharing a kernel). --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
In twinkle, try to use audio codec gsm. It is the most compressed out of all the audio codecs. I also have a ekiga account and want to try it with twinkle. If you have problems of bandwidth, then gsm should help. In most cases 128kbps upload is very problematic. On Thu, Feb 26, 2009 at 12:45 PM, Amos Shapira amos.shap...@gmail.comwrote: 2009/2/26 Tzafrir Cohen tzaf...@cohens.org.il: 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. That's exactly what I tried. I don't know how to configure asterisk to have a simple echo test working. Just a few minutes ago I called ekiga.net from Twinkle and for the first time I heard the instructions played by the echo test (I couldn't get even Ekiga itself to do that). But they are VERY chopy. Probably wrong audio device but only ALSA/default is accepted by Twinkle (not pulse) option. Do you have a simple asterisk echo test setup I can use? And how to configure Twinkle to connect to such a server? --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Some information that may be useful if anyone is still interested: - 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. - 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. - 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. Ori Berger wrote: FWIW, I asked DIDWW about the disappearing numbers (03 numbers are not there either at this moment), and they replied both 077 and 03 be back shortly (but I don't know what shortly means). Diamondcard still offers 03 numbers for $10/year. (Do note that their DID costs are competitive, but their termination rates are not). Israeli Numbers appear and disappear -- apparently they do not replenish their inventory regularly anymore, and you can get a number when someone gives their one up. However, if you want to use DIDWW -- talk to their sales team, they may have a DID for you even if it's not visible in the online shop. A US number is cheaper at diacmondcard. Israeli numbers were more available at DIDWW a month ago. Ori. ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
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. The only instance where I got incoming Audio was with Ekiga calling to sip:5...@ekiga.net (the ekiga echo test service) which allowed me to hear myself at about 4 second delay. Even their callback service (sip:5...@ekiga.net) kept saying that my SIP client rejected their callback. I use an ADSL2+ with D-Link DSL-G604T modem/router. I also have a Sipura ATA-3000 connected to my ISP's VoIP (SIP) service with no problem (and no port forwarding required in the modem). I can't configure the ATA to use other SIP servers because it supports registration to only one SIP server at a time and I don't want to loose access to the ISP's VoIP service. What SIP software do others use here that works for them? Does it support STUN? Ekiga's config droid says I need to use STUN (so I use stun.ekiga.net). Can anyone help me troubleshoot the audio problems? Ultimately, I'd like to connect Nokia E71 via SIP to the asterisk server, but that's another complexity I'll wait with until I get Asterisk working with a software SIP client. Thanks, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Thu, Feb 26, 2009 at 03:21:54PM +1100, Amos Shapira wrote: Why do you think OpenVZ might be better? Does it have a known advantage in Asterisk hosting? Because your provider may be able to provide you basically the same resources for less money. So if OpenVZ is good enough for you: fine. Oh, and as I mentioned non-dahdi (non-zaptel) timing for Asterisk, see the following for more information: http://lists.digium.com/pipermail/asterisk-dev/2009-February/036659.html -- 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
Re: Setting up a PBX for Israel-US communication
On Sat, Feb 14, 2009 at 08:05:06AM +1100, Amos Shapira wrote: 2009/2/14 Tzafrir Cohen tzaf...@cohens.org.il: On Fri, Feb 13, 2009 at 01:54:53PM +1100, Amos Shapira wrote: I saw somewhere that the Xen hosts provided by VPSLink already have 1000HTz clocks on them, saving a kernel recompilation. But is it actually 1000Hz? How can you tell without access to the kernel config? The CPU MHz is 2200. The experiment below was for sampling the clock Asterisk would get Try: Build DAHDI[*] on that system without installing it: svn co http://svn.digium.com/svn/dahdi/linux dahdi-linux svn co http://svn.digium.com/svn/dahdi/tools dahdi-tools Grr Make those two: svn co http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux svn co http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools cd dahdi-linux make # maybe you also need KVERS, KSRC and such cd ../dahdi-tools configure --with-dahdi=../dahdi-linux make dahdi_test modprobe crc_ccitt insmod ../dahdi-linux/drivers/dahdi/dahdi.ko insmod ../dahdi-linux/drivers/dahdi/dahdi_dummy.ko # and now finally test: ./dahdi_test -v -c 5 # When you're done: rmmod dahdi_dummy dahdi Anyway, just did that in a Xen host I have (using Debian Lenny) The Xen kernel does not have CONFIG_HIGH_RES_TIMERS set, I could not find a working RTC driver, and HZ is set to 250 . 1. Debian :-) (As I package Asterisk for it) I used Debian for over 10 years but now I got used to CentOS (simply because it's so much easier to find hosts which support it for my work needs). I recently looked for unmanaged hosts and there Debian was generally as common as Centos. Most managed hosts used cPanel and alike that I simply can't stand. I can't stand cPanel either. With Debian hosting at least on one place (I think Spry, the parent of VPSLink) I got stuck with an old Debian on a Virtuozo VPS which I can't upgrade without just installing the machine from scratch (I know Debian supports in-place upgrades, but the virtual host setup won't allow this). This is why I go for unmanaged. The box from which I'm writing this got upgraded from Etch to Lenny. Generally each Debian system supports the kernel of the previous distribution. 2. Do consider Asterisk SVN trunk. At least if this is a home PBX. Specifically The extra timing source stuff might be of direct interest to you. I don't know Asterisk at all so not sure I should try the latest and greatest where I'll never know whether something is broken because I did it wrong or because it's really broken. In that case, latest stable is now branch 1.6.0 (e.g. 1.6.0.5). It was 1.6rc1 That's 1.6.1-rc1 . That is: a release candidate for the branch 1.6.1 . A while before that 1.6.0.5 was released, which is the fifth maintinance release of branch 1.6.0 . about three days ago when I looked . I'd rather stick to something which reached .23 for now. 1.4 is still being maintained, but not sure for how long. 1.6 was rc1 just this week. Are you saying that once I decide to go with Asterisk I also have to keep close chase of their latest release in order to have it supported (i.e. bug and security fixes)? The closest thing I found for describing it is: http://www.asterisk.org/node/48539 At the moment 1.4 is still supported, but I'm not sure I'd use it for new installations. -- 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
Re: Setting up a PBX for Israel-US communication
2009/2/14 Tzafrir Cohen tzaf...@cohens.org.il: On Sat, Feb 14, 2009 at 08:05:06AM +1100, Amos Shapira wrote: 2009/2/14 Tzafrir Cohen tzaf...@cohens.org.il: On Fri, Feb 13, 2009 at 01:54:53PM +1100, Amos Shapira wrote: I saw somewhere that the Xen hosts provided by VPSLink already have 1000HTz clocks on them, saving a kernel recompilation. But is it actually 1000Hz? How can you tell without access to the kernel config? The CPU MHz is 2200. The experiment below was for sampling the clock Asterisk would get OK, I didn't realize that. Will try it later then. Anyway, just did that in a Xen host I have (using Debian Lenny) The Xen kernel does not have CONFIG_HIGH_RES_TIMERS set, I could not find a working RTC driver, and HZ is set to 250 . It doesn't necessarily mean that it's the same on my Xen host. As far as I follow it's a per-kernel-compilation setting. 1. Debian :-) (As I package Asterisk for it) I used Debian for over 10 years but now I got used to CentOS (simply because it's so much easier to find hosts which support it for my work needs). I recently looked for unmanaged hosts and there Debian was generally as common as Centos. Most managed hosts used cPanel and alike that I simply can't stand. I can't stand cPanel either. With Debian hosting at least on one place (I think Spry, the parent of VPSLink) I got stuck with an old Debian on a Virtuozo VPS which I can't upgrade without just installing the machine from scratch (I know Debian supports in-place upgrades, but the virtual host setup won't allow this). This is why I go for unmanaged. The box from which I'm writing this got upgraded from Etch to Lenny. Generally each Debian system supports the kernel of the previous distribution. What do you call unmanaged? It's a Virtuozo-style host, so the kernel is dictated by the hosting env. The VPSLink service I have is Xen so although they control the kernel I can pick one of few and probably switch to a new one. It was 1.6rc1 That's 1.6.1-rc1 . That is: a release candidate for the branch 1.6.1 . A while before that 1.6.0.5 was released, which is the fifth maintinance release of branch 1.6.0 . OK, I missed that. about three days ago when I looked . I'd rather stick to something which reached .23 for now. 1.4 is still being maintained, but not sure for how long. 1.6 was rc1 just this week. Are you saying that once I decide to go with Asterisk I also have to keep close chase of their latest release in order to have it supported (i.e. bug and security fixes)? The closest thing I found for describing it is: http://www.asterisk.org/node/48539 At the moment 1.4 is still supported, but I'm not sure I'd use it for new installations. Thanks for the pointer. It looks like the relevant sentence was cut in the middle: With Asterisk 1.4, once Asterisk 1.4.N is released, Asterisk 1.4.X is no longer supported, where X http://downloads.digium.com/pub/telephony/asterisk/asterisk-1.6.0.tar.gz; ? --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
BTW - About Israeli DID's - I just received an ad from Gizmo5 (I probably registered with them a while ago) advertising their new free SIP-to-Skype gateway. So I went to their web site and found that they offer a callin number for $12/3-months or $35/12-months, which is very close to the lost 077 numbers for $3/month with DIDWW. These are 03 numbers (http://gizmo5.com/pc/network/callin-numbers/). They explicitly support SIP and Asterisk (see the FAQ: http://support.gizmoproject.com/index.php?_m=knowledgebase_a=viewarticlekbarticleid=325nav=0). Does anyone have experience with them? Thanks, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Amos Shapira wrote: which is very close to the lost 077 numbers for $3/month with DIDWW. These are 03 numbers (http://gizmo5.com/pc/network/callin-numbers/). FWIW, I asked DIDWW about the disappearing numbers (03 numbers are not there either at this moment), and they replied both 077 and 03 be back shortly (but I don't know what shortly means). Diamondcard still offers 03 numbers for $10/year. (Do note that their DID costs are competitive, but their termination rates are not). ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Sat, Feb 14, 2009 at 09:04:08PM +1100, Amos Shapira wrote: 2009/2/14 Tzafrir Cohen tzaf...@cohens.org.il: On Sat, Feb 14, 2009 at 08:05:06AM +1100, Amos Shapira wrote: 2009/2/14 Tzafrir Cohen tzaf...@cohens.org.il: On Fri, Feb 13, 2009 at 01:54:53PM +1100, Amos Shapira wrote: I saw somewhere that the Xen hosts provided by VPSLink already have 1000HTz clocks on them, saving a kernel recompilation. But is it actually 1000Hz? How can you tell without access to the kernel config? The CPU MHz is 2200. The experiment below was for sampling the clock Asterisk would get OK, I didn't realize that. Will try it later then. Anyway, just did that in a Xen host I have (using Debian Lenny) The Xen kernel does not have CONFIG_HIGH_RES_TIMERS set, I could not find a working RTC driver, and HZ is set to 250 . It doesn't necessarily mean that it's the same on my Xen host. As far as I follow it's a per-kernel-compilation setting. 1. Debian :-) (As I package Asterisk for it) I used Debian for over 10 years but now I got used to CentOS (simply because it's so much easier to find hosts which support it for my work needs). I recently looked for unmanaged hosts and there Debian was generally as common as Centos. Most managed hosts used cPanel and alike that I simply can't stand. I can't stand cPanel either. With Debian hosting at least on one place (I think Spry, the parent of VPSLink) I got stuck with an old Debian on a Virtuozo VPS which I can't upgrade without just installing the machine from scratch (I know Debian supports in-place upgrades, but the virtual host setup won't allow this). This is why I go for unmanaged. The box from which I'm writing this got upgraded from Etch to Lenny. Generally each Debian system supports the kernel of the previous distribution. What do you call unmanaged? It's a Virtuozo-style host, so the kernel is dictated by the hosting env. The VPSLink service I have is Xen so although they control the kernel I can pick one of few and probably switch to a new one. It was 1.6rc1 That's 1.6.1-rc1 . That is: a release candidate for the branch 1.6.1 . A while before that 1.6.0.5 was released, which is the fifth maintinance release of branch 1.6.0 . OK, I missed that. about three days ago when I looked . I'd rather stick to something which reached .23 for now. 1.4 is still being maintained, but not sure for how long. 1.6 was rc1 just this week. Are you saying that once I decide to go with Asterisk I also have to keep close chase of their latest release in order to have it supported (i.e. bug and security fixes)? The closest thing I found for describing it is: http://www.asterisk.org/node/48539 At the moment 1.4 is still supported, but I'm not sure I'd use it for new installations. Thanks for the pointer. It looks like the relevant sentence was cut in the middle: With Asterisk 1.4, once Asterisk 1.4.N is released, Asterisk 1.4.X is no longer supported, where X http://downloads.digium.com/pub/telephony/asterisk/asterisk-1.6.0.tar.gz; ? Since then 1.4.23 has been published, and I saw a number of fixes scheduled for 1.4.24 . -- 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
Re: Setting up a PBX for Israel-US communication
On Fri, Feb 13, 2009 at 01:54:53PM +1100, Amos Shapira wrote: 2009/2/13 Tzafrir Cohen tzaf...@cohens.org.il: As I think I have mentioned before, I'm not sure how well it works under Xen. E.g. http://wiki.vpslink.com/Compiling_ztdummy_on_Xen has a considerable ammount of unneeded voodoo. The basic thing they do there is to use the internal timing intead of RTC. But do they actually have HZ set to 1000 on that system? Odd settings for such a machine. I saw somewhere that the Xen hosts provided by VPSLink already have 1000HTz clocks on them, saving a kernel recompilation. But is it actually 1000Hz? Try: Build DAHDI[*] on that system without installing it: svn co http://svn.digium.com/svn/dahdi/linux dahdi-linux svn co http://svn.digium.com/svn/dahdi/tools dahdi-tools cd dahdi-linux make # maybe you also need KVERS, KSRC and such cd ../dahdi-tools configure --with-dahdi=../dahdi-linux make dahdi_test modprobe crc_ccitt insmod ../dahdi-linux/drivers/dahdi/dahdi.ko insmod ../dahdi-linux/drivers/dahdi/dahdi_dummy.ko # and now finally test: ./dahdi_test -v -c 5 # When you're done: rmmod dahdi_dummy dahdi [*] I use DAHDI in this case because for Zaptel you'd still have to install udev rules, whereas DAHDI uses '!' in device names to make the device files be generated with proper names. This is handy if you don't have either installed and didn't set up udev rules yet. In addition to that, if you need to mess with kernel module building, you'll find that with Debian things actually work, whereas with Centos you have to fight the system harder to make them so. See also http://docs.tzafrir.org.il/#_kernel_configuration I would also recommend: 1. Debian :-) (As I package Asterisk for it) I used Debian for over 10 years but now I got used to CentOS (simply because it's so much easier to find hosts which support it for my work needs). I recently looked for unmanaged hosts and there Debian was generally as common as Centos. Most managed hosts used cPanel and alike that I simply can't stand. 2. Do consider Asterisk SVN trunk. At least if this is a home PBX. Specifically The extra timing source stuff might be of direct interest to you. I don't know Asterisk at all so not sure I should try the latest and greatest where I'll never know whether something is broken because I did it wrong or because it's really broken. In that case, latest stable is now branch 1.6.0 (e.g. 1.6.0.5). 1.4 is still being maintained, but not sure for how long. -- 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
Re: Setting up a PBX for Israel-US communication
On Fri, Feb 13, 2009 at 02:01:26PM +1100, Amos Shapira wrote: 2009/2/13 Ori Berger linux...@orib.net: Does that include the zaptel ztdummy server required for MeetMe and MusicOnHold? I couldn't get it to work; I don't really need it either, but it would be nice to have. I compiled the zaptel package without a problem but didn't install it. Which specific Zaptel package? Zaptel/DAHDI are not getting into Fedora because of RH's position regarding out-of-tree modules. As a result, there's not even single naming convention for packages of Zaptel and Zaptel modules. Quite a number of the existing ones are rather broken. -- 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
Re: Setting up a PBX for Israel-US communication
2009/2/14 Tzafrir Cohen tzaf...@cohens.org.il: On Fri, Feb 13, 2009 at 01:54:53PM +1100, Amos Shapira wrote: I saw somewhere that the Xen hosts provided by VPSLink already have 1000HTz clocks on them, saving a kernel recompilation. But is it actually 1000Hz? How can you tell without access to the kernel config? The CPU MHz is 2200. Try: Build DAHDI[*] on that system without installing it: svn co http://svn.digium.com/svn/dahdi/linux dahdi-linux svn co http://svn.digium.com/svn/dahdi/tools dahdi-tools cd dahdi-linux make # maybe you also need KVERS, KSRC and such cd ../dahdi-tools configure --with-dahdi=../dahdi-linux make dahdi_test modprobe crc_ccitt insmod ../dahdi-linux/drivers/dahdi/dahdi.ko insmod ../dahdi-linux/drivers/dahdi/dahdi_dummy.ko # and now finally test: ./dahdi_test -v -c 5 # When you're done: rmmod dahdi_dummy dahdi [*] I use DAHDI in this case because for Zaptel you'd still have to install udev rules, whereas DAHDI uses '!' in device names to make the device files be generated with proper names. This is handy if you don't have either installed and didn't set up udev rules yet. Thanks for the pointer. But do I need all this? I got the impression that if I don't use meetme (what's that for?) or another feature which people manage by without then I don't need all this at all. In addition to that, if you need to mess with kernel module building, you'll find that with Debian things actually work, whereas with Centos you have to fight the system harder to make them so. I'll see. Thanks. When I start using this knowledge for my workplace I'll have to do it on CentOS anyway (it's our SOE now, by my own decision) so I'll have to deal with it anyway. See also http://docs.tzafrir.org.il/#_kernel_configuration I would also recommend: 1. Debian :-) (As I package Asterisk for it) I used Debian for over 10 years but now I got used to CentOS (simply because it's so much easier to find hosts which support it for my work needs). I recently looked for unmanaged hosts and there Debian was generally as common as Centos. Most managed hosts used cPanel and alike that I simply can't stand. I can't stand cPanel either. With Debian hosting at least on one place (I think Spry, the parent of VPSLink) I got stuck with an old Debian on a Virtuozo VPS which I can't upgrade without just installing the machine from scratch (I know Debian supports in-place upgrades, but the virtual host setup won't allow this). 2. Do consider Asterisk SVN trunk. At least if this is a home PBX. Specifically The extra timing source stuff might be of direct interest to you. I don't know Asterisk at all so not sure I should try the latest and greatest where I'll never know whether something is broken because I did it wrong or because it's really broken. In that case, latest stable is now branch 1.6.0 (e.g. 1.6.0.5). It was 1.6rc1 about three days ago when I looked . I'd rather stick to something which reached .23 for now. 1.4 is still being maintained, but not sure for how long. 1.6 was rc1 just this week. Are you saying that once I decide to go with Asterisk I also have to keep close chase of their latest release in order to have it supported (i.e. bug and security fixes)? Cheers, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
2009/2/14 Tzafrir Cohen tzaf...@cohens.org.il: On Fri, Feb 13, 2009 at 02:01:26PM +1100, Amos Shapira wrote: 2009/2/13 Ori Berger linux...@orib.net: Does that include the zaptel ztdummy server required for MeetMe and MusicOnHold? I couldn't get it to work; I don't really need it either, but it would be nice to have. I compiled the zaptel package without a problem but didn't install it. Which specific Zaptel package? I followed the instructions from http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation which points to http://downloads.digium.com/pub/zaptel/zaptel-1.4-current.tar.gz Zaptel/DAHDI are not getting into Fedora because of RH's position regarding out-of-tree modules. As a result, there's not even single naming convention for packages of Zaptel and Zaptel modules. Quite a number of the existing ones are rather broken. Are you refering to Zaptel RPM packages? I compiled from source. I generally try to avoid installing just any rpm out there in the wild. --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
OK, So I've setup CentOS 5.2 on Xen VPS at VPSLink, compiled latest Asterisk (1.4.23) very smoothly according to the instructions at http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation up to and including the asterisk -...vvvc and stop now. Also installed the rest of the addons. Only deviation I see is that I setup MySQL not to listen on TCP sockets. Other then that I copied all the config bits from that page to the corresponding files. Now what? How do I test this? I have a Nokia E71 with a built-in SIP client which I'd like to connect to this thing. I'd like to have Israeli, a Brazilian and possibly temporarily an Australian number which will ring on my phone. Going to didww.com I'm not sure what should I look for - Phone to VOIP or Phone to IP-PBX? both options cost $US10 a month, I don't see an option to pick the allegedly cheaper 077 numbers. Anything beyond about $5/month makes this possibly uneconomical, as for the long term I don't spend that much on international calls and Skypeout subscription can provide unlimited calls for 5 euro/month (for minimum of three months). (We have 4000 free Skype minutes from our mobiles so Skypeout is very convenient to call from wherever we are). Thanks, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
2009/2/12 Amos Shapira amos.shap...@gmail.com: How do I test this? Write an extensions file and use http://www.didww.com/service_did.php to test DIDs for free. I have a Nokia E71 with a built-in SIP client which I'd like to connect to this thing. Set up the credentials in sip.conf and connect from the Nokia, verify you can register and you can see the registration. Going to didww.com I'm not sure what should I look for - Phone to VOIP or Phone to IP-PBX? both options cost $US10 a month, I don't see an option to pick the allegedly cheaper 077 numbers. Indeed, it is gone from their screen. A mistake perhaps? Try emailing sa...@didww.com. Disclaimer: I'm not affiliated with DIDWW in any way other than being a happy customer. Anything beyond about $5/month makes this possibly uneconomical, as for the long term I don't spend that much on international calls and Skypeout subscription can provide unlimited calls for 5 euro/month (for minimum of three months). (We have 4000 free Skype minutes from our mobiles so Skypeout is very convenient to call from wherever we are). For me it's not about my cost, it's about the (perceived) cost of people who call me. This way I can have people call an Israeli number to get at me and they know they don't pay much. Plus don't dis the geek factor... -- Arik ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Thu, Feb 12, 2009 at 11:04:07PM +1100, Amos Shapira wrote: I have a Nokia E71 with a built-in SIP client which I'd like to connect to this thing. The easiest way to test it is to use a computer with a SIP client. Then you have access to network debugging tools, etc. From Windows and Mac I prefer X-Lite which is free as in beer, not open source. http://www.counterpath.net/x-lite.htmlactive=4 There are plenty of SIP clients for Linux. I'd like to have Israeli, a Brazilian and possibly temporarily an Australian number which will ring on my phone. Anything beyond about $5/month makes this possibly uneconomical, as for the long term I don't spend that much on international calls and Skypeout subscription can provide unlimited calls for 5 euro/month (for minimum of three months). (We have 4000 free Skype minutes from our mobiles so Skypeout is very convenient to call from wherever we are). Skype is a cheap, but IMHO not very good alternative. From my experience it's not consistent. Sometimes it's good, sometimes it's unusable, often it is just ok. Considering their price $6 (US) or $13 (US) with more countries it's a good deal, especially because they are up front on what too much is. Last I checked there was no SkypeIn from Israel, although it is now one of their unlimited SkypeOut countries. Although I expect that most of the people reading this are too young to remember satellite long distance calls, most of the time, it's better than they were. If you make a lot of calls to Israeli cell phones, you might want to check out Orange's deal. It's 600 minutes a month of outgoing calls to any number in Israel, plus the ISP side of a 2.5m line for 139 NIS. Additional minutes are 29ag, which is the cheapest I've seen to a cell phone, but awfully high to a landline. I don't know if they would let you take it and just use time with your own IP PBX, or if you would have to add your own system after their box. Since if I remember correctly, you don't live here, you would be in effect buying a family member an internet connection. Geoff. -- Geoffrey S. Mendelson, Jerusalem, Israel g...@mendelson.com N3OWJ/4X1GM ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Amos Shapira wrote: OK, So I've setup CentOS 5.2 on Xen VPS at VPSLink, compiled latest Asterisk (1.4.23) very smoothly according to the instructions at http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation up to and including the asterisk -...vvvc and stop now. Also installed the rest of the addons. Does that include the zaptel ztdummy server required for MeetMe and MusicOnHold? I couldn't get it to work; I don't really need it either, but it would be nice to have. Lylix.net is priced competitively with VPSlink, and have ztdummy available within their virtual machines - however, their lowest level is equivalent to VPSlink's link2 which sets you back at $15/month. I'm considering switching over there if I'd need meetme and didn't get ztdummy to work on VPSlink. How do I test this? I recommend twinkle on Linux. It has a log view that is immensely helpful in debugging, and got me much farther than Ekiga or Wengo did. Going to didww.com I'm not sure what should I look for - Phone to VOIP or Phone to IP-PBX? both options cost $US10 a month, I don't see an option to pick the allegedly cheaper 077 numbers. There were also 073 numbers (cellcom) that were $3/month. However, everything except 03 disappeared from DIDWW and also diamondcard.us - this doesn't look like a coincidence. Perhaps someone knows what has happened? I already have a couple of 077 DIDs but this is troubling. Anything beyond about $5/month makes this possibly uneconomical, as for the long term I don't spend that much on international calls and Skypeout subscription can provide unlimited calls for 5 euro/month (for minimum of three months). (We have 4000 free Skype minutes from our mobiles so Skypeout is very convenient to call from wherever we are). Having played with Asterisk a little, and having set up hosting and stuff - I'd say that there are services that will be pricewise competitive, such as Jajah, Skype, and OlehPhone. Each one has a different set of restrictions on the flexibility that an asterisk server provides, but they take away all the headaches. If you're only looking to save money, this is probably NOT the best option when you also factor in the cost of your time, paid or leisure. ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Thu, Feb 12, 2009 at 06:15:50PM +0200, Ori Berger wrote: Amos Shapira wrote: OK, So I've setup CentOS 5.2 on Xen VPS at VPSLink, compiled latest Asterisk (1.4.23) very smoothly according to the instructions at http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation up to and including the asterisk -...vvvc and stop now. Also installed the rest of the addons. Does that include the zaptel ztdummy server required for MeetMe and MusicOnHold? I couldn't get it to work; I don't really need it either, but it would be nice to have. As I think I have mentioned before, I'm not sure how well it works under Xen. E.g. http://wiki.vpslink.com/Compiling_ztdummy_on_Xen has a considerable ammount of unneeded voodoo. The basic thing they do there is to use the internal timing intead of RTC. But do they actually have HZ set to 1000 on that system? Odd settings for such a machine. See also http://docs.tzafrir.org.il/#_kernel_configuration I would also recommend: 1. Debian :-) (As I package Asterisk for it) 2. Do consider Asterisk SVN trunk. At least if this is a home PBX. Specifically The extra timing source stuff might be of direct interest to you. -- 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
Re: Setting up a PBX for Israel-US communication
2009/2/13 Tzafrir Cohen tzaf...@cohens.org.il: As I think I have mentioned before, I'm not sure how well it works under Xen. E.g. http://wiki.vpslink.com/Compiling_ztdummy_on_Xen has a considerable ammount of unneeded voodoo. The basic thing they do there is to use the internal timing intead of RTC. But do they actually have HZ set to 1000 on that system? Odd settings for such a machine. I saw somewhere that the Xen hosts provided by VPSLink already have 1000HTz clocks on them, saving a kernel recompilation. See also http://docs.tzafrir.org.il/#_kernel_configuration I would also recommend: 1. Debian :-) (As I package Asterisk for it) I used Debian for over 10 years but now I got used to CentOS (simply because it's so much easier to find hosts which support it for my work needs). 2. Do consider Asterisk SVN trunk. At least if this is a home PBX. Specifically The extra timing source stuff might be of direct interest to you. I don't know Asterisk at all so not sure I should try the latest and greatest where I'll never know whether something is broken because I did it wrong or because it's really broken. Thanks. --Amos PS - My main quest now (apart from actually getting to call through my new asterisk server) is to find a reasonably priced Israeli DID. So far they all seem to be around $US9.95 and above. ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
2009/2/13 Ori Berger linux...@orib.net: Does that include the zaptel ztdummy server required for MeetMe and MusicOnHold? I couldn't get it to work; I don't really need it either, but it would be nice to have. I compiled the zaptel package without a problem but didn't install it. Lylix.net is priced competitively with VPSlink, and have ztdummy available within their virtual machines - however, their lowest level is equivalent to VPSlink's link2 which sets you back at $15/month. I'm considering switching over there if I'd need meetme and didn't get ztdummy to work on VPSlink. How do I test this? I recommend twinkle on Linux. It has a log view that is immensely helpful in debugging, and got me much farther than Ekiga or Wengo did. Thanks. Will try that. Going to didww.com I'm not sure what should I look for - Phone to VOIP or Phone to IP-PBX? both options cost $US10 a month, I don't see an option to pick the allegedly cheaper 077 numbers. There were also 073 numbers (cellcom) that were $3/month. However, everything except 03 disappeared from DIDWW and also diamondcard.us - this doesn't look like a coincidence. Perhaps someone knows what has happened? I already have a couple of 077 DIDs but this is troubling. If anyone hears about good place to get Israeli DID's then please speak up. Is there any chance that the phone companies themselves will talk to me with reasonable prices if I approach them directly? Which department would that be? Having played with Asterisk a little, and having set up hosting and stuff - I'd say that there are services that will be pricewise competitive, such as Jajah, Skype, and OlehPhone. Each one has a different set of restrictions on Jajah is getting more and more expensive to use for us. Skypeout looks OK on price up to a point. Olephone, if that's that one I remember from a while ago, looked too expensive for my needs. the flexibility that an asterisk server provides, but they take away all the headaches. If you're only looking to save money, this is probably NOT the best option when you also factor in the cost of your time, paid or leisure. As much as I'm busy, I see this not just as a way to cut costs but also as an investment since I will be called to setup company-wide voip at some stage when we get around to it (actually be in charge with, but it's good to know what your workers are up to). Cheers, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
I've been using Asterisk for the last 4 years. Never had a reason to move to Freeswitch. Asterisk rarely crash, (we have more than 15M minutes) If you like perl so much (or any language for that matter) you could configure Asterisk dialplan using a script instead of asterisk regular dialplans... if you are used to unix style configuration files, asterisk is not that hard, and you could always use one of the web interfaces. I personally, really likes the ruby on rails intergartaion to asterisk, makes it a lot of fun. just my 2cents. Ohad On Wed, Feb 11, 2009 at 1:12 PM, Amos Shapira amos.shap...@gmail.comwrote: 2009/2/10 Tzafrir Cohen tzaf...@cohens.org.il: FreeSwitch developers have many bad words regarding Asterisk. So many of them are unfounded (or no longer founded) that I generally distrust them. This guy claims to be within the first tier of Asterisk commiters and to know its code through and through. Has Asterisk managed to get rid of the deadlocks and segfaults (and apparently a prehistoric architecture) he mentions in his FreeSwitch vs Asterisk at http://freeswitch.org/node/117? At the moment Asterisk is more mature and far more deployed. So it appears. But also I keep hearing horror stories about configuring it, and the guy who mentioned FreeSWITCH in the link from my previous message had experience with Asterisk and prefers FreeSWITCH. What merit points are there for Asterisk beyond everyone uses it (a billion flies CAN be wrong, you know)? Can it do something that FreeSwitch can't (the FreeSwitch guy says something about Asterisk being a PBX while FreeSwitch is a software switch, I don't know what's the difference and for now I plan to use it only for myself and maybe to connect a couple of trans-pacific offices)? (and does support Lua, BTW. Only nobody really bothers using it. As the fact that most people didn't touch the pbx_perl and pbx_js that the author of FS wrote as Asterisk modules before starting FS) Anyway, FS's license is MPL. Which for me is a concern to avoid using it: yet another GPL-incompatible software does not help anybody. In all the arguments above I didn't see one which actually refers to the merits of FreeSwitch. I'm not trying to annoy, just understand what am I missing about it, if at all. So far your points against it are not conclusive, IMHO. Thanks, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
2009/2/11 Ohad Levy ohadl...@gmail.com: I've been using Asterisk for the last 4 years. Never had a reason to move to Freeswitch. Asterisk rarely crash, (we have more than 15M minutes) Thanks very much for your input. rarely crashes sounds a bit wacky but 15 million minutes is impressive. I think I'll start with Asterisk just to see that I can get it running (and connect my mobile to it as a SIP client) and then see whether I want to switch/try FreeSwitch. If you like perl so much (or any language for that matter) you could configure Asterisk dialplan using a script instead of asterisk regular I don't know anything about Asterisk so can't decide whether I want to use a script yet. My only experience with a dialplan was on my Sipura 3000 and I even lost my password to that one (will have to reset it eventually). dialplans... if you are used to unix style configuration files, asterisk is not that hard, and you could always use one of the web interfaces. I just stumbled upon FreePBX. Is it worth anything? Cheers, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Amos Shapira wrote: 2009/2/11 Ohad Levy ohadl...@gmail.com: I've been using Asterisk for the last 4 years. Never had a reason to move to Freeswitch. Asterisk rarely crash, (we have more than 15M minutes) Asterisk runs Codefidence IPBX for more then 5 years. Never seen a crash. Asterisk runs the Tel Aviv branch of the Israeli Law bar Odphone system(Lishkat Orchey Ha Dim Mahuz TA) for more then an year now. The only two reported bugs were hardware (firmware update from Sangoma fixed that ) and our own software :-) I and others can go on. Asterisk works. I donno about the other stuff (FreePBX etc.) but Asterisk is prodction level system *If you know what you are doing*. Being that you are an experienced Linux user that has no issue editing rc files or RTFM I believe you've already got 95% of the reasons why Asterisk installation fail covered. :-) Gilad -- Gilad Ben-Yossef Chief Coffee Drinker Codefidence Ltd. The code is free, your time isn't.(TM) Web:http://codefidence.com Email: gi...@codefidence.com Office: +972-8-9316883 ext. 201 Fax:+972-8-9316885 Mobile: +972-52-8260388 The Doctor: Don't worry, Reinette, just a nightmare. Everyone has nightmares. Even monsters from under the bed have nightmares, don't you, monster? Reinette: What do monsters have nightmares about? The Doctor: Me! ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
2009/2/11 Gilad Ben-Yossef gi...@codefidence.com: I and others can go on. Asterisk works. I donno about the other stuff Thanks for the testimonial. (FreePBX etc.) but Asterisk is prodction level system *If you know what you FreePBX claims to be a nice web gui for Asterisk, that's why I asked about it. are doing*. Being that you are an experienced Linux user that has no issue editing rc files or RTFM I believe you've already got 95% of the reasons why Asterisk installation fail covered. :-) Thanks for the reassurance. I've just ordered a vpslink Xen guest but the ssh latency from my home is very slow (I'm actually connected through a pretty good ISP). I'll try it from work tomorrow. Is 200ms ping time too much for SIP, or is it irrelevant. Cheers, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Amos Shapira wrote: 2009/2/11 Gilad Ben-Yossef gi...@codefidence.com: I and others can go on. Asterisk works. I donno about the other stuff Thanks for the testimonial. (FreePBX etc.) but Asterisk is prodction level system *If you know what you FreePBX claims to be a nice web gui for Asterisk, that's why I asked about it. are doing*. Being that you are an experienced Linux user that has no issue editing rc files or RTFM I believe you've already got 95% of the reasons why Asterisk installation fail covered. :-) Thanks for the reassurance. I've just ordered a vpslink Xen guest but the ssh latency from my home is very slow (I'm actually connected through a pretty good ISP). I'll try it from work tomorrow. Is 200ms ping time too much for SIP, or is it irrelevant. Very relevant. Callers usually notice roundtrip voice delays of 250ms or more.The recommendation for VoIP is a one way delay of less then 150ms. Anything more will become very noticeable in conversation and in real life 150ms is too high as well since jitter before induce around 10ms addtional latency. In your case, 200ms round trip time, assuming a symmetric distribution, of delay (but mind you that the internet isn't really symmetrical...) is 100ms one way delay so this is not excellent but not really bad either. In theory, at least, you shouldn't be able to notice this delay in conversation. Gilad -- Gilad Ben-Yossef Chief Coffee Drinker Codefidence Ltd. The code is free, your time isn't.(TM) Web:http://codefidence.com Email: gi...@codefidence.com Office: +972-8-9316883 ext. 201 Fax:+972-8-9316885 Mobile: +972-52-8260388 The Doctor: Don't worry, Reinette, just a nightmare. Everyone has nightmares. Even monsters from under the bed have nightmares, don't you, monster? Reinette: What do monsters have nightmares about? The Doctor: Me! ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Gilad Ben-Yossef wrote: Is 200ms ping time too much for SIP, or is it irrelevant. Very relevant. Callers usually notice roundtrip voice delays of 250ms or more.The recommendation for VoIP is a one way delay of less then 150ms. Anything more will become very noticeable in conversation and in real life 150ms is too high as well since jitter before induce around 10ms addtional latency. In your case, 200ms round trip time, assuming a symmetric distribution, of delay (but mind you that the internet isn't really symmetrical...) is 100ms one way delay so this is not excellent but not really bad either. In theory, at least, you shouldn't be able to notice this delay in conversation. Forgot to add - here is a great resournce for all things VoIP QoS: http://www.voip-info.org/wiki-QoS Gilad -- Gilad Ben-Yossef Chief Coffee Drinker Codefidence Ltd. The code is free, your time isn't.(TM) Web:http://codefidence.com Email: gi...@codefidence.com Office: +972-8-9316883 ext. 201 Fax:+972-8-9316885 Mobile: +972-52-8260388 The Doctor: Don't worry, Reinette, just a nightmare. Everyone has nightmares. Even monsters from under the bed have nightmares, don't you, monster? Reinette: What do monsters have nightmares about? The Doctor: Me! ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Wed, Feb 11, 2009 at 01:38:39PM +0200, Gilad Ben-Yossef wrote: In your case, 200ms round trip time, assuming a symmetric distribution, of delay (but mind you that the internet isn't really symmetrical...) is 100ms one way delay so this is not excellent but not really bad either. In theory, at least, you shouldn't be able to notice this delay in conversation. Since a VoIP conversation is two way, even if the latency is not symetrical, one side or the other will experience the higher of the two sides. So while peak latency would be better a metric, IMHO ping time is a decent average. Geoff. -- Geoffrey S. Mendelson, Jerusalem, Israel g...@mendelson.com N3OWJ/4X1GM ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Gilad Ben-Yossef wrote: Amos Shapira wrote: Is 200ms ping time too much for SIP, or is it irrelevant. Very relevant. Callers usually notice roundtrip voice delays of 250ms or more.The recommendation for VoIP is a one way delay of less then 150ms. Anything more will become very noticeable in conversation and in real life 150ms is too high as well since jitter before induce around 10ms addtional latency. While it's not ideal and can be noticeable, it's not as bad as all that. I regularly speak with people in Australia which is over 400 ms ping time away, and while we occasionally talk over each other, it really doesn't impact much at all. I'd imagine it could be a problem if you made it much bigger though. Geoff. ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
2009/2/11 Geoffrey S. Mendelson g...@mendelson.com: So while peak latency would be better a metric, IMHO ping time is a decent average. Thanks Geoff and Aviram for the replies and pointers. I'll try to see how I measure the latency (maybe iperf?). Cheers, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Wed, Feb 11, 2009 at 10:21:20PM +1100, Amos Shapira wrote: 2009/2/11 Gilad Ben-Yossef gi...@codefidence.com: I and others can go on. Asterisk works. I donno about the other stuff Thanks for the testimonial. (FreePBX etc.) but Asterisk is prodction level system *If you know what you FreePBX claims to be a nice web gui for Asterisk, that's why I asked about it. are doing*. Being that you are an experienced Linux user that has no issue editing rc files or RTFM I believe you've already got 95% of the reasons why Asterisk installation fail covered. :-) Thanks for the reassurance. I've just ordered a vpslink Xen guest but the ssh latency from my home is very slow (I'm actually connected through a pretty good ISP). I'll try it from work tomorrow. Is 200ms ping time too much for SIP, or is it irrelevant. Besides the latency, another factor to consider is the jitter: $ ping -c 10 kernel.org PING kernel.org (204.152.191.37) 56(84) bytes of data. 64 bytes from pub2.kernel.org (204.152.191.37): icmp_seq=1 ttl=52 time=232 ms 64 bytes from pub2.kernel.org (204.152.191.37): icmp_seq=2 ttl=52 time=232 ms 64 bytes from pub2.kernel.org (204.152.191.37): icmp_seq=3 ttl=52 time=233 ms 64 bytes from pub2.kernel.org (204.152.191.37): icmp_seq=4 ttl=52 time=233 ms 64 bytes from pub2.kernel.org (204.152.191.37): icmp_seq=5 ttl=52 time=233 ms 64 bytes from pub2.kernel.org (204.152.191.37): icmp_seq=6 ttl=52 time=232 ms 64 bytes from pub2.kernel.org (204.152.191.37): icmp_seq=7 ttl=52 time=233 ms 64 bytes from pub2.kernel.org (204.152.191.37): icmp_seq=8 ttl=52 time=233 ms 64 bytes from pub2.kernel.org (204.152.191.37): icmp_seq=9 ttl=52 time=232 ms 64 bytes from pub2.kernel.org (204.152.191.37): icmp_seq=10 ttl=52 time=232 ms --- kernel.org ping statistics --- 10 packets transmitted, 10 received, 0% packet loss, time 9038ms rtt min/avg/max/mdev = 232.102/233.035/233.960/0.773 ms As you can see, I have a very predictable latency. This generally means packets will arrive on time. Here's a less optimal example: $ ping -c 10 lxer.linux.no ping: unknown host lxer.linux.no tzaf...@sweetmorn:~$ ping -c 10 lxr.linux.no PING lxr.linpro.no (87.238.46.5) 56(84) bytes of data. 64 bytes from lxr.linpro.no (87.238.46.5): icmp_seq=1 ttl=45 time=235 ms 64 bytes from lxr.linpro.no (87.238.46.5): icmp_seq=2 ttl=45 time=203 ms 64 bytes from lxr.linpro.no (87.238.46.5): icmp_seq=3 ttl=45 time=211 ms 64 bytes from lxr.linpro.no (87.238.46.5): icmp_seq=4 ttl=45 time=210 ms 64 bytes from lxr.linpro.no (87.238.46.5): icmp_seq=5 ttl=45 time=205 ms 64 bytes from lxr.linpro.no (87.238.46.5): icmp_seq=6 ttl=45 time=209 ms 64 bytes from lxr.linpro.no (87.238.46.5): icmp_seq=7 ttl=45 time=208 ms 64 bytes from lxr.linpro.no (87.238.46.5): icmp_seq=8 ttl=45 time=207 ms 64 bytes from lxr.linpro.no (87.238.46.5): icmp_seq=9 ttl=45 time=200 ms 64 bytes from lxr.linpro.no (87.238.46.5): icmp_seq=10 ttl=45 time=242 ms --- lxr.linpro.no ping statistics --- 10 packets transmitted, 10 received, 0% packet loss, time 9038ms rtt min/avg/max/mdev = 200.634/213.566/242.968/13.399 ms -- 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
Re: Setting up a PBX for Israel-US communication
On Tue, Feb 10, 2009 at 03:44:14PM +1100, Amos Shapira wrote: I have yet to take up Arik Baratz on his generous offer to help me setup my own Asterisk server, but in the meantime I learne a couple of things: 1. Amazon EC2 might contain surprises in the bill. I'm saying this VERY cautiously based on one test server we had for a few days with lots of disk space (it runs a web application with a 300Gb disk for MySQL database) but it should be checked carefully before jumping on it. 2. The following is a thread on the other mailing list I follow where people give some more useful advise about setting up VoIP networks http://lists.slug.org.au/archives/slug/2009/02/msg00051.html. In particular I'm curios to hear what others have to say about FreeSwitch (freeswitch.org) compared to Asterisk, as I keep hearing that Asterisk configuration is somewhat of a black art (if it turns to be as much black art as Perl programming then I might even enjoy it :) and the guy here http://lists.slug.org.au/archives/slug/2009/02/msg00059.html who raised the FreeSWITCH option says that nothing less than a gun pointed at his head will make him go back to Asterisk. FreeSwitch developers have many bad words regarding Asterisk. So many of them are unfounded (or no longer founded) that I generally distrust them. At the moment Asterisk is more mature and far more deployed. (and does support Lua, BTW. Only nobody really bothers using it. As the fact that most people didn't touch the pbx_perl and pbx_js that the author of FS wrote as Asterisk modules before starting FS) Anyway, FS's license is MPL. Which for me is a concern to avoid using it: yet another GPL-incompatible software does not help anybody. -- 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
Re: Setting up a PBX for Israel-US communication
2009/2/10 Tzafrir Cohen tzaf...@cohens.org.il: FreeSwitch developers have many bad words regarding Asterisk. So many of them are unfounded (or no longer founded) that I generally distrust them. This guy claims to be within the first tier of Asterisk commiters and to know its code through and through. Has Asterisk managed to get rid of the deadlocks and segfaults (and apparently a prehistoric architecture) he mentions in his FreeSwitch vs Asterisk at http://freeswitch.org/node/117? At the moment Asterisk is more mature and far more deployed. So it appears. But also I keep hearing horror stories about configuring it, and the guy who mentioned FreeSWITCH in the link from my previous message had experience with Asterisk and prefers FreeSWITCH. What merit points are there for Asterisk beyond everyone uses it (a billion flies CAN be wrong, you know)? Can it do something that FreeSwitch can't (the FreeSwitch guy says something about Asterisk being a PBX while FreeSwitch is a software switch, I don't know what's the difference and for now I plan to use it only for myself and maybe to connect a couple of trans-pacific offices)? (and does support Lua, BTW. Only nobody really bothers using it. As the fact that most people didn't touch the pbx_perl and pbx_js that the author of FS wrote as Asterisk modules before starting FS) Anyway, FS's license is MPL. Which for me is a concern to avoid using it: yet another GPL-incompatible software does not help anybody. In all the arguments above I didn't see one which actually refers to the merits of FreeSwitch. I'm not trying to annoy, just understand what am I missing about it, if at all. So far your points against it are not conclusive, IMHO. Thanks, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
I have yet to take up Arik Baratz on his generous offer to help me setup my own Asterisk server, but in the meantime I learne a couple of things: 1. Amazon EC2 might contain surprises in the bill. I'm saying this VERY cautiously based on one test server we had for a few days with lots of disk space (it runs a web application with a 300Gb disk for MySQL database) but it should be checked carefully before jumping on it. 2. The following is a thread on the other mailing list I follow where people give some more useful advise about setting up VoIP networks http://lists.slug.org.au/archives/slug/2009/02/msg00051.html. In particular I'm curios to hear what others have to say about FreeSwitch (freeswitch.org) compared to Asterisk, as I keep hearing that Asterisk configuration is somewhat of a black art (if it turns to be as much black art as Perl programming then I might even enjoy it :) and the guy here http://lists.slug.org.au/archives/slug/2009/02/msg00059.html who raised the FreeSWITCH option says that nothing less than a gun pointed at his head will make him go back to Asterisk. Cheers, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
(I contacted several people off list for personal requests, I'm moving the discussion back on-list because I think I saw interest in it expressed before). Someone basically said I'll help you if you get yourself a server. I like to keep costs down, I like highly-available servers (doing this 24/7 for my workplace now), and I started playing with Amazon EC2 for work too - 1+1=... :) Here are a few links I found in a quick search about running Asterisk on Amazon EC2: http://www.scribd.com/doc/3905321/PREVIEW-CloudCrunch-Howto-Asterisk-PBX-and-Amazon-EC2 And an interesting collection by Nir Simionovich, which contains a link to asterisk.org.il (according to NoScript): http://www.simionovich.com/?tag=asterisk Including his tales of setting up stuff on EC2. Hope someone finds this useful, and would like to cooperate with on setting something like this up. Cheers, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Sat, Jan 31, 2009 at 5:49 AM, Ori Berger linux...@orib.net wrote: Least year, there was a thread discussing set up of an asterisk system, which included a description by Arik Baratz (see e.g. http://www.mail-archive.com/linux-il@cs.huji.ac.il/msg52213.html and http://www.mail-archive.com/linux-il@cs.huji.ac.il/msg52276.html) I am planning to set up something similar, and before I embark, wanted to ask anyone on the list if they have anything to add to that discussion. From looking around, it seems that: - VPSLink is still the cheapest VPS host at $8/month (or $80/year) for 64MB of memory. It seems like the OpenVZ package is better suited than the Xen package, being less resource intensive. And from past experience I would bet on Debian -- however, can anyone here share their experience (Arik?). Will apt-get install asterisk be enough, or will I have to compile everything myself? If you are only going to install it for VoIP without any PRI/FXO support then yes, if you require to also provide support for PRI/FXO/FXS then you will require to compile the dahadi/zaptel drivers (they are kernel drivers so m-a can help you with it after using make menu to choose what to compile. - grnvoip still seems like the cheapest termination service - but only provides SIP connection, whereas voipjet, still competitive, provides only IAX2. Any recommendation here? IAX2 is supposed to be less resource intensive than SIP, but I don't know if that'll matter on a 64MB machine routing at most two calls. IAX2 require only one UDP port, while SIP requires a big range of ports to be open. However you can use tunneling with SIP, so it will work only with one port (good for NAT and firewalls). Personally I prefer IAX2, because it is less complex and less security hell, but they both doing the work well. - didww.com is competitive on DIDs ($3/month for 077- number in IL, $10/month for 03- number, $2/month US number), but other such as diamondcard.us provide same prices, and also do termination (although not as cheaply as grnvoip or voipjet). - Any positive or negative experiences routing SMS between those systems? Does anyone have experience, specific software versions and/or configuration scripts to share with regards to such a setup? Thanks in advance, Ori. Ido ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On Fri, Jan 30, 2009 at 10:49:34PM -0500, Ori Berger wrote: Least year, there was a thread discussing set up of an asterisk system, which included a description by Arik Baratz (see e.g. http://www.mail-archive.com/linux-il@cs.huji.ac.il/msg52213.html and http://www.mail-archive.com/linux-il@cs.huji.ac.il/msg52276.html) I am planning to set up something similar, and before I embark, wanted to ask anyone on the list if they have anything to add to that discussion. From looking around, it seems that: - VPSLink is still the cheapest VPS host at $8/month (or $80/year) for 64MB of memory. It seems like the OpenVZ package is better suited than the Xen package, being less resource intensive. And from past experience I would bet on Debian -- however, can anyone here share their experience (Arik?). Will apt-get install asterisk be enough, or will I have to compile everything myself? The package in Lenny, yes. While there are no official backports for Etch, I have some unofficial ones. But then again, if you have a new system at this stage, Lenny is something to consider. I'm not sure that this package will provide you enough resources if you intend to use more than a minimal installation (more than 1-2 concurrent calls), assuming calls do use some compressed codec. - grnvoip still seems like the cheapest termination service - but only provides SIP connection, whereas voipjet, still competitive, provides only IAX2. Any recommendation here? IAX2 is supposed to be less resource intensive than SIP, but I don't know if that'll matter on a 64MB machine routing at most two calls. Not really. -- 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
Re: Setting up a PBX for Israel-US communication
On 31/01/2009, at 05:49, Ori Berger wrote: - VPSLink is still the cheapest VPS host at $8/month (or $80/year) for 64MB of memory. It seems like the OpenVZ package is better suited than the Xen package, being less resource intensive. And from past experience I would bet on Debian -- however, can anyone here share their experience (Arik?). Will apt-get install asterisk be enough, or will I have to compile everything myself? Worse than that, asterisk will not work in an OpenVZ VE unless you have access to the underlying host to install the zaptel kernel modules. Does anyone have experience, specific software versions and/or configuration scripts to share with regards to such a setup? The truth is, between the cost of a VPS and termination of calls, I would think you're better off just signing up with a voip provider. Can you do better than $25/month for unlimited calling to the US? (Disclaimer: I am the CTO of a US VoIP provider who offers Israeli DIDs and calling plans as well. Note that I am not hawking my company's services in this email! If you're curious what we offer, contact me off-list.) --sambo ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
sammy ominsky wrote: Worse than that, asterisk will not work in an OpenVZ VE unless you have access to the underlying host to install the zaptel kernel modules. Is this the case now? I know is used to be the case that you only needed zaptel if you were going to use a device that needed it or you needed the timing it provided (e.g. to run a conference bridge). I have it running on a machine running Debian Etch and it's not running Zaptel and in fact doesn't even list it as a dependency. Of course it is running Asterisk 1.2.13 so this may not apply to newer versions. Though it also occurs to me that we're running 1.4.18.1 on a Xen'based virtual host and, apart from the lack of MeetMe, is also running fine without Zaptel. Geoff. ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
On 31/01/2009, at 23:12, Geoff Shang wrote: Worse than that, asterisk will not work in an OpenVZ VE unless you have access to the underlying host to install the zaptel kernel modules. Is this the case now? I know is used to be the case that you only needed zaptel if you were going to use a device that needed it or you needed the timing it provided (e.g. to run a conference bridge). I actually meant to go back and insert the word properly between work and in, got distracted by my daughter, and forgot. I don't have any asterisk instances running without a timing source, so none of my attempts at OpebnVZ would work. If you don't need the zaptel modules or ztdummy for timing, there's probably no reason it wouldn't work. But that also means no call waiting, no putting calls on hold, no channels or spans... honestly, it leaves you with a pretty limited system in my opinion. If all you need is the ability to place a call, though, it's fine. --sambo ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
sammy ominsky wrote: Worse than that, asterisk will not work in an OpenVZ VE unless you have access to the underlying host to install the zaptel kernel modules. (Note that in another email, Sammy mentions that it works but some features don't). It looks like Xen would therefore be needed? The truth is, between the cost of a VPS and termination of calls, I would think you're better off just signing up with a voip provider. Can you do better than $25/month for unlimited calling to the US? For the kind of setup I want (similar to what Arik described in his first email), no one offers that at all, let alone at $25. The most important feature being accessible from ANY phone (e.g., us mobile to il mobile at local prices, and the other way around, without having any predefined list of destination on any side) without any FXO/FXS or other equipment needed. But it's not about the money -- primarily, it is because I _want_ to tinker with Asterisk in the process, and trying to minimize the budget at the same time just adds a little spice. Thanks, Ori. ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
2009/2/1 Ori Berger linux...@orib.net: sammy ominsky wrote: Worse than that, asterisk will not work in an OpenVZ VE unless you have access to the underlying host to install the zaptel kernel modules. (Note that in another email, Sammy mentions that it works but some features don't). It looks like Xen would therefore be needed? Personally I'm using OpenVZ. I wanted to switch to Xen, but didn't put the time and effort into it. I get what I need from the system, and yes it does complain that it doesn't have a timing source, but It Works For Me (tm). -- Arik ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: Setting up a PBX for Israel-US communication
Hey Ori, long time. 2009/1/31 Ori Berger linux...@orib.net: - VPSLink is still the cheapest VPS host at $8/month (or $80/year) for 64MB of memory. It seems like the OpenVZ package is better suited than the Xen package, being less resource intensive. And from past experience I would bet on Debian -- however, can anyone here share their experience (Arik?). Will apt-get install asterisk be enough, or will I have to compile everything myself? I have installed Ubuntu and not Debian. I installed Asterisk from packages, I didn't compile anything. In fact I don't have any dev tools in my machine and I doubt they will run with only 64MB of RAM. Heck I have to stop Asterisk when I want to run some commands, like for example apt-get... - grnvoip still seems like the cheapest termination service - but only provides SIP connection, whereas voipjet, still competitive, provides only IAX2. Any recommendation here? IAX2 is supposed to be less resource intensive than SIP, but I don't know if that'll matter on a 64MB machine routing at most two calls. I use voipjet/IAX2. Viopjet claim that they are not to be used by end users, and I simply ignore that. So far I haven't asked for support and haven't gotten any. They have the occasional downtime, if you use a DNS name for the host and not an IP you will usually not feel it because they change DNS records to compensate. You have to have more than $20 in your account at all times or else you can't use most of their servers. - didww.com is competitive on DIDs ($3/month for 077- number in IL, $10/month for 03- number, $2/month US number), but other such as diamondcard.us provide same prices, and also do termination (although not as cheaply as grnvoip or voipjet). I use didww.com. I did not check out any others. I have a number in the US, in Israel and in Australia. I used to have a number in France but some stupid French decided to limit VoIP numbers to the physical region they seem to be from, so lacking an address in Paris I had to give that number up. - Any positive or negative experiences routing SMS between those systems? Didn't try it, I have no idea if it will be successful. I know Nir Simionovich and Oded Arbel have messed around with SMS quite a bit, and I think they are both on the list. Does anyone have experience, specific software versions and/or configuration scripts to share with regards to such a setup? I can share my extensions.conf with you if you want. -- Arik ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Setting up a PBX for Israel-US communication
Least year, there was a thread discussing set up of an asterisk system, which included a description by Arik Baratz (see e.g. http://www.mail-archive.com/linux-il@cs.huji.ac.il/msg52213.html and http://www.mail-archive.com/linux-il@cs.huji.ac.il/msg52276.html) I am planning to set up something similar, and before I embark, wanted to ask anyone on the list if they have anything to add to that discussion. From looking around, it seems that: - VPSLink is still the cheapest VPS host at $8/month (or $80/year) for 64MB of memory. It seems like the OpenVZ package is better suited than the Xen package, being less resource intensive. And from past experience I would bet on Debian -- however, can anyone here share their experience (Arik?). Will apt-get install asterisk be enough, or will I have to compile everything myself? - grnvoip still seems like the cheapest termination service - but only provides SIP connection, whereas voipjet, still competitive, provides only IAX2. Any recommendation here? IAX2 is supposed to be less resource intensive than SIP, but I don't know if that'll matter on a 64MB machine routing at most two calls. - didww.com is competitive on DIDs ($3/month for 077- number in IL, $10/month for 03- number, $2/month US number), but other such as diamondcard.us provide same prices, and also do termination (although not as cheaply as grnvoip or voipjet). - Any positive or negative experiences routing SMS between those systems? Does anyone have experience, specific software versions and/or configuration scripts to share with regards to such a setup? Thanks in advance, Ori. ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
