Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 07, 2009 at 01:47:57PM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 10:09 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk. Your quote This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls. Isn't that more or less the same thing I said that you snipped, Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. And what if the call does not go through a TDM card? And ore importantly: how can you tell who is the caller and who is the callee? The rtp-tap interface basically tells you that channel X had a call at time Y. I am sure it is pretty trivial to figure out who channel X and Y are based on the channel, time, CID, DID Just a wee bit of code... Which means you have to keep a separate DB of that (I know such DB exists: the CDR) and get that data from it. Extra work to do. Some people prefer to avoid it. If it does not go through a TDM card, and is VoIP, then port mirroring works just fine. Sipcallid is a very simple way to match callers to callees. VoIP mirroring implies you have control over the network infrastructure. What if you install the PBX in a hostile network where the network administrator doesn't like you sniffing other network traffic? Not to mention that it is extra setup. So we add a different option. One that depends on Asterisk sending the relevant data, and uses the existing monitoring infrastructure in Asterisk: simply use Monitor and StopMonitor to enable/disable monitoring. This is something Asterisk admins should be familiar with. I snip content that is not relevant to my reply. Whoever reads this list already read about the Sangoma interface previously. I had nothing to say about it. It was not related to that new branch. Not everyone who reads the list, reads all the posts, give me a break. It was related to the thread. My target audince in posts to asterisk-users is (surpirse-surpirse) the readers of asterisk-users. I generally do expect them to follow the list[1]. Your motives and alliances have and always will be for Xorcom and Digium. That is the only reason why you helped me with that BRI install in the US, so you could poke around
Re: [asterisk-users] Using asterisk as the recording server
using mixmonitor might not be such a good idea. afaik the mixing of the recordings of the two channels starts after ending the call causing a high cpu load. if you have recordings going on all the time moving the 2 files that has to be mixed to a dedicated mixing server might be a good idea. after mixing it should be stored in a retrievable way. Erik de Wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 8 sep 2009 om 00:25 heeft Miguel Molina mmol...@millenium.com.co het volgende geschreven:\ I imagine this setup will need those two communicating entities to be part of the pabx. But support extension 100 of PABX A (legacy) calls 101 on the same platform. I want asterisk connected to PABX A via E1/T1 to know about that call and start recording (tap) without bridging or being part of that conversation Hi, Asterisk won't work as a recording server if the call doesn't go through it. In the IP world it means that both media (RTP) and signalling must pass through asterisk, and in the E1/T1 digital or analog world it means that the call must be bridged through asterisk. A simple dialplan would explain it: exten = s,1,Answer() ;Asterisk receives the call, from the lecagy PBX or from the external link (this should be two different contexts) exten = s,n,MixMonitor(blah) ; Records the conversation, exten = s,n,Dial(Tech/peer/number...,30,rtTwhatever) ; and sends the call back to the legacy PBX or to an external link If you want to record 100% calls, you would have to route every call through asterisk, even internal PBX calls. Even if you want to tap your legacy PBX to a non-asterisk recording server like the ones suggested before in this thread, the calls must go through a link to make tapping possible and you should seek an alternate solution to the internal calls within your legacy PBX. The beauty of asterisk and open source IP-PBXs relies on the native recording capabilities which makes things really easy. When you see that asterisk works and that can do the recordings and much more, you would start thinking on making asterisk your main PBX solution and leaving that legacy PBX for minimal uses. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
Again, how many calls were you able record using RAMdisk? Anywhere 300? As I stated before, this is going to be dependent on how you're manipulating the calls and the gear you're running on. The nice thing about your 'just broadcast the entire LAN to the recording solution' is that the recording service just gets to throw away everything that's not an audio channel, and it doesn't have to do squat to the call. If it COULDN'T do a lot of recordings under these circumstances it wouldn't be worth any money. I don't think I've pushed my solution past 90 simultaneous recordings of MeetMe() mixing, with more than 100 AGI channels running, with assorted ChanSpy() jobs. Bookmark my post, so when you reach your RAMDisk limit, you can join the big league. Anything I do as a scaling solution will be price versus performance. So since we're talking about a commercial solution to replace something that asterisk does, I'll have to find out what your commercial solution costs per channel, and compare that against the cost of cloning out an identical server. My solution scales to parallel servers just fine. Is OrecX really $199 per recorded channel? So that 300 channels you're talking about costs $60,000? So I can buy six $10,000 servers, each of which can run circles around my current solution, and still break even. I like my solution better. OrecX has a free version. I guess you didin't really check it out since your mind was already made up. 300+ Simultaneous calls recorded in perfect clarity for the price of an R200 or if you want higher end, a DL360 Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Tue, Sep 8, 2009 at 5:08 AM, Erik de Wildi...@meetmecall.nl wrote: using mixmonitor might not be such a good idea. afaik the mixing of the recordings of the two channels starts after ending the call causing a high cpu load. Incorrect. The 'mix' in Mixmonitor() is that two legs of a call are mixed together as the recording happens. It is Monitor() that works in the way you describe, although 'high cpu load' is dependent on how you mix. It would be 'bursty' cpu load if you choose to mix as soon as a recording completes. Since Mixmonitor() is mixing as the recording happens, there is instantly a retrievable recording when the recording completes. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
I imagine this setup will need those two communicating entities to be part of the pabx. But support extension 100 of PABX A (legacy) calls 101 on the same platform. I want asterisk connected to PABX A via E1/T1 to know about that call and start recording (tap) without bridging or being part of that conversation Hi, Asterisk won't work as a recording server if the call doesn't go through it. In the IP world it means that both media (RTP) and signalling must pass through asterisk, and in the E1/T1 digital or analog world it means that the call must be bridged through asterisk. A simple dialplan would explain it: exten = s,1,Answer() ;Asterisk receives the call, from the lecagy PBX or from the external link (this should be two different contexts) exten = s,n,MixMonitor(blah) ; Records the conversation, exten = s,n,Dial(Tech/peer/number...,30,rtTwhatever) ; and sends the call back to the legacy PBX or to an external link If you want to record 100% calls, you would have to route every call through asterisk, even internal PBX calls. Even if you want to tap your legacy PBX to a non-asterisk recording server like the ones suggested before in this thread, the calls must go through a link to make tapping possible and you should seek an alternate solution to the internal calls within your legacy PBX. The beauty of asterisk and open source IP-PBXs relies on the native recording capabilities which makes things really easy. When you see that asterisk works and that can do the recordings and much more, you would start thinking on making asterisk your main PBX solution and leaving that legacy PBX for minimal uses. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center Erik de Wild escribió: using mixmonitor might not be such a good idea. afaik the mixing of the recordings of the two channels starts after ending the call causing a high cpu load. if you have recordings going on all the time moving the 2 files that has to be mixed to a dedicated mixing server might be a good idea. after mixing it should be stored in a retrievable way. No, that was the old behavior of Monitor() with the m option that at the end of the recording it launched an underneath sox process which did the mix, causing a CPU spike on every conversation end and putting asterisk on trouble if there were many mixes at the same time. Mixmonitor took care of that, and it does the mixing while the conversation is taking place, thus generating the single file with no CPU spikes or external process calls. Your idea about the separate mixing server was what our company did about three years ago with the old first 1.2 asterisk versions, where MixMonitor used to be buggy and we were forced to implement that kind of solution. But times are a lot better now! Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk. Your quote This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls. Isn't that more or less the same thing I said that you snipped, Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. This isn't the biz list, nor the dev list. Snipping out the reference of Sangoma being able to do RTP tap and suggesting people use your experimental dev branch doesn't really help users very much. I really enjoy your use of selective snipping, quoting, and taking things out of context to manipulate threads. You should be a reporter. Too bad it doesn't work on me and I will call you out on it. Please let us users know when your branch gets merged into a Stable Release -- Senior Systems and Network Administrator Triple Canopy, Inc., 2250 Corporate Park Drive, Suite 300 ph. +1.703.673.5191 mob.+1.240.938.1212 FAX.+1.703.673.1279 steve.tot...@triplecanopy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk. Your quote This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls. Isn't that more or less the same thing I said that you snipped, Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. And what if the call does not go through a TDM card? And ore importantly: how can you tell who is the caller and who is the callee? The rtp-tap interface basically tells you that channel X had a call at time Y. If you control recording through the monitoring interface of Asterisk you can start and stop the recording when you need it. You can also provide better information aobut the call. But again, it means that this is part of Asterisk, and I figure Sangoma has quite a few non-Asterisk customers. This isn't the biz list, nor the dev list. Snipping out the reference of Sangoma being able to do RTP tap and suggesting people use your experimental dev branch doesn't really help users very much. My message was an explicit call for testers, if you haven't noticed :-) I snip content that is not relevant to my reply. Whoever reads this list already read about the Sangoma interface previously. I had nothing to say about it. It was not related to that new branch. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 7, 2009 at 10:09 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk. Your quote This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls. Isn't that more or less the same thing I said that you snipped, Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. And what if the call does not go through a TDM card? And ore importantly: how can you tell who is the caller and who is the callee? The rtp-tap interface basically tells you that channel X had a call at time Y. I am sure it is pretty trivial to figure out who channel X and Y are based on the channel, time, CID, DID Just a wee bit of code... If it does not go through a TDM card, and is VoIP, then port mirroring works just fine. Sipcallid is a very simple way to match callers to callees. If you control recording through the monitoring interface of Asterisk you can start and stop the recording when you need it. You can also provide better information aobut the call. But again, it means that this is part of Asterisk, and I figure Sangoma has quite a few non-Asterisk customers. Sounds neat, when will it be out of beta? This isn't the biz list, nor the dev list. Snipping out the reference of Sangoma being able to do RTP tap and suggesting people use your experimental dev branch doesn't really help users very much. My message was an explicit call for testers, if you haven't noticed :-) I snip content that is not relevant to my reply. Whoever reads this list already read about the Sangoma interface previously. I had nothing to say about it. It was not related to that new branch. Not everyone who reads the list, reads all the posts, give me a break. It was related to the thread. Your motives and alliances have and always will be for Xorcom and Digium. That is the only reason why you helped me with that BRI install in the US, so you could poke around and try to figure out how Marcin Pycko achieved what you cannot. I may check it out when it is part of a stable backported to 1.4 release, otherwise, I don't run beta in production. Sometimes large sums of money rely on systems, as do much more valuable human lives. -- Senior Systems and Network Administrator Triple Canopy, Inc., 2250 Corporate Park Drive, Suite 300 ph.
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk. Your quote This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls. Isn't that more or less the same thing I said that you snipped, Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. And what if the call does not go through a TDM card? And ore importantly: how can you tell who is the caller and who is the callee? The rtp-tap interface basically tells you that channel X had a call at time Y. If you control recording through the monitoring interface of Asterisk you can start and stop the recording when you need it. You can also provide better information aobut the call. But again, it means that this is part of Asterisk, and I figure Sangoma has quite a few non-Asterisk customers. This isn't the biz list, nor the dev list. Snipping out the reference of Sangoma being able to do RTP tap and suggesting people use your experimental dev branch doesn't really help users very much. My message was an explicit call for testers, if you haven't noticed :-) I snip content that is not relevant to my reply. Whoever reads this list already read about the Sangoma interface previously. I had nothing to say about it. It was not related to that new branch. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir I imagine this setup will need those two communicating entities to be part of the pabx. But support extension 100 of PABX A (legacy) calls 101 on the same platform. I want asterisk connected to PABX A via E1/T1 to know about that call and start recording (tap) without bridging or being part of that conversation ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
I imagine this setup will need those two communicating entities to be part of the pabx. But support extension 100 of PABX A (legacy) calls 101 on the same platform. I want asterisk connected to PABX A via E1/T1 to know about that call and start recording (tap) without bridging or being part of that conversation Hi, Asterisk won't work as a recording server if the call doesn't go through it. In the IP world it means that both media (RTP) and signalling must pass through asterisk, and in the E1/T1 digital or analog world it means that the call must be bridged through asterisk. A simple dialplan would explain it: exten = s,1,Answer() ;Asterisk receives the call, from the lecagy PBX or from the external link (this should be two different contexts) exten = s,n,MixMonitor(blah) ; Records the conversation, exten = s,n,Dial(Tech/peer/number...,30,rtTwhatever) ; and sends the call back to the legacy PBX or to an external link If you want to record 100% calls, you would have to route every call through asterisk, even internal PBX calls. Even if you want to tap your legacy PBX to a non-asterisk recording server like the ones suggested before in this thread, the calls must go through a link to make tapping possible and you should seek an alternate solution to the internal calls within your legacy PBX. The beauty of asterisk and open source IP-PBXs relies on the native recording capabilities which makes things really easy. When you see that asterisk works and that can do the recordings and much more, you would start thinking on making asterisk your main PBX solution and leaving that legacy PBX for minimal uses. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 7, 2009 at 12:29 AM, Steve Totarostot...@first-notification.com wrote: Did you push it past 300 on two year old hardware and software? old hardware yes. old software no. The servers are more than 3 years old Core 2 Duo Dell Dimension desktop as proof of concept? are core 2 duo's really two years old already? I guess so. I don't really follow the latest hardware news. I have my lab on server-class gear. Port mirroring is basic on almost any newer switch. Login, enable port monitoring, write mem, done. Port mirroring is basic on quality networking gear. I know perfectly well how it works. My point was that replicating ALL traffic on a LAN port seemed a bit like hauling out all the corn plants from the corn field when what you really wanted was just the corn kernels from the ears. That's what I mean by heavy-handed. I've never used the software you've proposed. I realize that replicating all traffic for a port, or in my case, all traffic for a bonded interface is not difficult logically, and is quick to configure. I think it is aesthetically displeasing compared to grabbing the recordings at the place where the calls are already taking place. Personal taste. You're allowed your opinion too, which you've clearly stated. I build robust and redundant systems, separate server for DB, recording, gateways, in an all HA configuration. Me too. Again, taste. Again, how many calls were you able record using RAMdisk? Anywhere 300? As I stated before, this is going to be dependent on how you're manipulating the calls and the gear you're running on. The nice thing about your 'just broadcast the entire LAN to the recording solution' is that the recording service just gets to throw away everything that's not an audio channel, and it doesn't have to do squat to the call. If it COULDN'T do a lot of recordings under these circumstances it wouldn't be worth any money. I don't think I've pushed my solution past 90 simultaneous recordings of MeetMe() mixing, with more than 100 AGI channels running, with assorted ChanSpy() jobs. Bookmark my post, so when you reach your RAMDisk limit, you can join the big league. Anything I do as a scaling solution will be price versus performance. So since we're talking about a commercial solution to replace something that asterisk does, I'll have to find out what your commercial solution costs per channel, and compare that against the cost of cloning out an identical server. My solution scales to parallel servers just fine. Is OrecX really $199 per recorded channel? So that 300 channels you're talking about costs $60,000? So I can buy six $10,000 servers, each of which can run circles around my current solution, and still break even. I like my solution better. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using asterisk as the recording server
Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. http://wiki.sangoma.com/wanpipe-voice-rtp-tap -- Senior Systems and Network Administrator Triple Canopy, Inc., 2250 Corporate Park Drive, Suite 300 ph. +1.703.673.5191 mob.+1.240.938.1212 FAX.+1.703.673.1279 steve.tot...@triplecanopy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
I have also seen: PSTN asterisk legacy Which also gives you a migration path PaulH Research wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
Paul, He already said, While am aware and active user of astersk monitor function for recording so I don't think migration path is an issue. A dedicated recording server is recommended if you are going to be recording a good deal of calls. You certainly would not want to run out of hard drive space on your Asterisk server and bring it down. Also, with Asterisk (last I knew) ~60 simultaneous calls, the audio starts breaking up very badly due to I/O. OrecX can do over 300 simultaneous calls and only need port mirroring enabled on your switch. Even if it crashes or HD fills, call go on normally. I have coined the term Passive Recording since the recording process does not touch Asterisk in any way, shape, or form. Senior Systems and Network Administrator Triple Canopy, Inc., 2250 Corporate Park Drive, Suite 300 ph. +1.703.673.5191 mob.+1.240.938.1212 FAX.+1.703.673.1279 steve.tot...@triplecanopy.com On Sun, Sep 6, 2009 at 11:21 PM, Paul Hales pdha...@optusnet.com.au wrote: I have also seen: PSTN asterisk legacy Which also gives you a migration path PaulH Research wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Sun, Sep 6, 2009 at 11:46 PM, Steve Totarostot...@asteriskhelpdesk.com wrote: A dedicated recording server is recommended if you are going to be recording a good deal of calls. You certainly would not want to run out of hard drive space on your Asterisk server and bring it down. Bring it down, really? I think monitor would just complain that it couldn't write to a device. I suppose you could have problems if your recording partition was also your system partition, but that would be true for any application, such as apache web activity logs. Also, with Asterisk (last I knew) ~60 simultaneous calls, the audio starts breaking up very badly due to I/O. This would be channel and system independent. For instance i/o blocking could cause problems but why would it affect simple non-mixed audio, like simple bridged Dahdi channels? OrecX can do over 300 simultaneous calls and only need port mirroring enabled on your switch. Even if it crashes or HD fills, call go on normally. If a non-system hd fills, calls will go on normally. Port mirroring seems like a pretty heavy-handed way to do call recording. How about asterisk, writing to a ramdisk for recordings, and every five minutes or so syncing off the completed recordings to a SAN? (You may have guessed I did this, and pushed it past 60 simultaneous recordings). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
Did you push it past 300 on two year old hardware and software? Core 2 Duo Dell Dimension desktop as proof of concept? Port mirroring is basic on almost any newer switch. Login, enable port monitoring, write mem, done. With a GUI, it takes all of thirty seconds. I don't see how this is heavy handed I build robust and redundant systems, separate server for DB, recording, gateways, in an all HA configuration. Again, how many calls were you able record using RAMdisk? Anywhere 300? Bookmark my post, so when you reach your RAMDisk limit, you can join the big league. Thanks, Steve Totaro On Mon, Sep 7, 2009 at 12:05 AM, David Backeberg dbackeb...@gmail.comwrote: On Sun, Sep 6, 2009 at 11:46 PM, Steve Totarostot...@asteriskhelpdesk.com wrote: A dedicated recording server is recommended if you are going to be recording a good deal of calls. You certainly would not want to run out of hard drive space on your Asterisk server and bring it down. Bring it down, really? I think monitor would just complain that it couldn't write to a device. I suppose you could have problems if your recording partition was also your system partition, but that would be true for any application, such as apache web activity logs. Also, with Asterisk (last I knew) ~60 simultaneous calls, the audio starts breaking up very badly due to I/O. This would be channel and system independent. For instance i/o blocking could cause problems but why would it affect simple non-mixed audio, like simple bridged Dahdi channels? OrecX can do over 300 simultaneous calls and only need port mirroring enabled on your switch. Even if it crashes or HD fills, call go on normally. If a non-system hd fills, calls will go on normally. Port mirroring seems like a pretty heavy-handed way to do call recording. How about asterisk, writing to a ramdisk for recordings, and every five minutes or so syncing off the completed recordings to a SAN? (You may have guessed I did this, and pushed it past 60 simultaneous recordings). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Senior Systems and Network Administrator Triple Canopy, Inc., 2250 Corporate Park Drive, Suite 300 ph. +1.703.673.5191 mob.+1.240.938.1212 FAX.+1.703.673.1279 steve.tot...@triplecanopy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
In layman's terms, if you read the OP's post, they want a *recording server*. With no mention of any other functionality. Asterisk *COULD* be used as a recording server but that is really not the proper tool, on the other hand OrecX *IS* a recording server, therefore the proper tool. I *COULD* use a butter knife for a screw driver, but the screw driver *IS*the proper tool for the job. Believe me, I have used the butter knife and it worked OK sometimes, other times it did not work at all, but the proper screwdriver did the job perfectly. I would dare say that using Asterisk simply for recording is heavy handed. Thanks, Steve Totaro On Mon, Sep 7, 2009 at 12:29 AM, Steve Totaro stot...@first-notification.com wrote: Did you push it past 300 on two year old hardware and software? Core 2 Duo Dell Dimension desktop as proof of concept? Port mirroring is basic on almost any newer switch. Login, enable port monitoring, write mem, done. With a GUI, it takes all of thirty seconds. I don't see how this is heavy handed I build robust and redundant systems, separate server for DB, recording, gateways, in an all HA configuration. Again, how many calls were you able record using RAMdisk? Anywhere 300? Bookmark my post, so when you reach your RAMDisk limit, you can join the big league. Thanks, Steve Totaro On Mon, Sep 7, 2009 at 12:05 AM, David Backeberg dbackeb...@gmail.comwrote: On Sun, Sep 6, 2009 at 11:46 PM, Steve Totarostot...@asteriskhelpdesk.com wrote: A dedicated recording server is recommended if you are going to be recording a good deal of calls. You certainly would not want to run out of hard drive space on your Asterisk server and bring it down. Bring it down, really? I think monitor would just complain that it couldn't write to a device. I suppose you could have problems if your recording partition was also your system partition, but that would be true for any application, such as apache web activity logs. Also, with Asterisk (last I knew) ~60 simultaneous calls, the audio starts breaking up very badly due to I/O. This would be channel and system independent. For instance i/o blocking could cause problems but why would it affect simple non-mixed audio, like simple bridged Dahdi channels? OrecX can do over 300 simultaneous calls and only need port mirroring enabled on your switch. Even if it crashes or HD fills, call go on normally. If a non-system hd fills, calls will go on normally. Port mirroring seems like a pretty heavy-handed way to do call recording. How about asterisk, writing to a ramdisk for recordings, and every five minutes or so syncing off the completed recordings to a SAN? (You may have guessed I did this, and pushed it past 60 simultaneous recordings). -- Senior Systems and Network Administrator Triple Canopy, Inc., 2250 Corporate Park Drive, Suite 300 ph. +1.703.673.5191 mob.+1.240.938.1212 FAX.+1.703.673.1279 steve.tot...@triplecanopy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. http://wiki.sangoma.com/wanpipe-voice-rtp-tap -- Senior Systems and Network Administrator Triple Canopy, Inc., 2250 Corporate Park Drive, Suite 300 ph. +1.703.673.5191 mob.+1.240.938.1212 FAX.+1.703.673.1279 steve.tot...@triplecanopy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
