Re: [asterisk-users] VOIP Monitoring tools........
Hi I use ADSL and SDSL on a lot of multi channel VoIP connections SIP and H323 and no real problems if you size the link correctly - this normally means limiting no of calls to match available bandwidth. Check out the upstream and downstream data rates and size on the smaller - normally the upstream. the thing that is not normally explained is the contention ratio at the local exchange. here in the UK residential broadband packages normally have a contention ratio of 50:1 so at peak you could be fighting with 49 other users. Contention ratios for business DSL packages can vary from 1:1 to typically 20:1. To monitor I use wireshark and look at packet loss on RTP. My home adsl connection is 7M downstream / 800K upstream with a 10:1 contention.I can normally get 20 x G729 channels without a problem much more and things get stressed - also I have to watch what other things are going on - downloads not too much of a problem because of the unbalanced downstream/upstream speeds but peer to peer stuff is a real killer! Paddy _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mike mosier Sent: 25 April 2010 02:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VOIP Monitoring tools Hey all What VoIP networking monitoring, asterisk monitoring tools would you recommend? I started working with an IT company that insists on using DSL with a Sonicwall router. The problem is that the clients are having sound problems. The owner is convinced that it's the Asterisk box. In the 4 yrs I have been doing this I have not had this bad a sound problem and it always came down to a bad setup in the Cisco router. Asterisk just doesn't have sound problems so I am going to have to convince him that its either the router or DSL. Has anyone used DSL for SIP traffic? How about Sonicwall routers? Michael D Mosier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jitter Buffer and MeetMe.
Hello, David. Thank you for reply. But my problem is certainly in the size of JitterBuffer of chan_local. I realy need to know how to change the size of JB (reduce). BTW: 1. The file /etc/asterisk/dsp.conf doesn't exist in my Asterisk 1.6.0.6 (something wrong?). 2. VAD is already disable for all trunks. 3. And 'talker optimization' future is already disable by patch for app_meetme.c. 2010/4/25 David Backeberg > On Fri, Apr 23, 2010 at 4:34 PM, russian qwerty > wrote: > > Hello. > > > > As I see, there are a lot of threads about jitter buffer... Maybe anybody > > knows something about my case? Any help will be appreciate. > > > > So, the problem with voice quality was completely solved, BUT some > customers > > have informed me about big latency. It's really hard to make dialogue > with > > current latency. > > You're on the right track here, but I don't think your problem is > jitter. I think your problem is VoIP and voice activity detection, and > depending on your version of asterisk, MeetMe conference 'talker > optimization'. > > I've posted all of this before. Here goes again... > > * 'talker optimization' should be disabled on MeetMe() conferences. > * /etc/asterisk/dsp.conf set silencethreshold=1024 > * /etc/asterisk/codecs.conf set vad=>false > > Give those a try, restart or reload asterisk to apply changes, and > tell us if it fixes it. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6, dialplans, and IVR
Hi, I have read the docs, and now I want to attempt to setup Asterisk 1.6. I am not going to complicate it with load balancing, etc. The setup is just 1 SIP line - no other in-house connections. All inbound traffic. I intend to keep this simple. Imagine that I sell pies in my neighborhood.. I want to make it easy to order them. So I create a dialplan to receive a call, record the caller id/ANI and time stamp, pickup the call, enter the IVR function ( play the "Press 1 for Apple, 2 for Cherry, 3 Blueberry"), get the button presses/input, and hangup. Later, I can get fancy, with order confirmation, etc. What I want to know is this: As inbound call volume increases, and my 1 server begins to struggle, do I: 1. Add a second server that just does IVR? Is that a very advanced AGI thing? 2. Add a second server that offloads other CPU intensive functions? 3. Just keep it simple, add a 2nd server that is a clone of the first, and load balance the calls - especially if you may bring UltraMonkey and *maybe* DRBD (shared disk/iSCSI) into the picture in the future? BTW, I am a Unix SA by trade, and so I am not a complete newbie. It seems like the easy answer is #3, ESPECIALLY if you are already busy with IT functions, plus baking the pies! Thanks very much in advance. Greg Banschbach -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to debug the problem of Asterisk using so much of CPU percentage...?
Hi Everyone, How is this possible? How can I go about debugging this? I think that the sound chopping and choking is also related to this. I have never seen Asterisk show 43% of cpu usuage when there is only one call going. It actually flactuates down to 11% and up to 43%. Please guide me as to where to look at? PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 2111 asterisk 17 0 40992 17m 8064 S 43.5 1.0 853:53.66 asterisk Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to debug the problem of Asterisk using so much of CPU percentage...?
Please take a few minutes, and fill us in on a few things: Which version of Asterisk, what codec if any, server hardware ( Make (HP/Dell/IBM) Model, CPU ( single or multicore, speed ), and any other pertinent info you can think of. Just trying to help you get a more informed answer. Good Luck, Greg From: bruce bruce To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sun, April 25, 2010 11:59:51 AM Subject: [asterisk-users] How to debug the problem of Asterisk using so much of CPU percentage...? Hi Everyone, How is this possible? How can I go about debugging this? I think that the sound chopping and choking is also related to this. I have never seen Asterisk show 43% of cpu usuage when there is only one call going. It actually flactuates down to 11% and up to 43%. Please guide me as to where to look at? PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 2111 asterisk 17 0 40992 17m 8064 S 43.5 1.0 853:53.66 asterisk Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Archlinux
I did use it for my first asterisk installation, but I moved to Debian due to things I disliked in arch as a server distro. I did not and still do not use Dahdi at all (http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge). I compiled everything from source without PKGBUILDS, since they are quite annoyting when talking about re-compiling of a package. On Sun, Apr 25, 2010 at 12:16 AM, ik wrote: > On Sat, Apr 24, 2010 at 21:19, Christian wrote: >> >> Hi all, >> Is anyone here using Asterisk on Archlinux? > > Yes and no, I do use it on Archlinux for testing purpose but not as a > server. > Arch linux is not built to be a server distro, unlike Debian that have extra > steps for process handling, like restarting a service that was just updated > and more. > >> >> If so, was it much to do in order for it to work? > > You need to either use the AUR builds or download the ABS information and > build it for yourself. Personally I use the yaourt tool as a pacman front > end. > >> >> Do you also use Dahdi? > > Like any other Asterisk it must have a dahdi module, at least dahdi_dummy. > >> >> many thanks, >> Christian >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > Ido > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP monitoring tools
Hi, On Sun, Apr 25, 2010 at 5:06 AM, mike mosier wrote: > Howdy all > > 1. does anyone know a good voip / sip / qos monitoring tool? Wireshark is quite good at it http://wiki.wireshark.org/VoIP_calls However I could only find it good for "debugging", not monitoring (tcpdump the whole RTP stream is too expensive for me:) Another notable piece of software (however proprietary and quite costly) is VQManager. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Michael Wegner
http://www.villasantilles.com/home.php -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI Congestion cause 34
Hi,
I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk and
try to make a call I get the following error message:
-- Executing [6781...@default:1] Dial("IAX2/iaxy-7477",
"DAHDI/g1/96781948") in new stack
[Apr 25 13:00:10] WARNING[3772]: app_dial.c:1806 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'IAX2/iaxy-7477' status is 'CONGESTION'
Restarting DAHDI does not help, for somehow after several hours the problem
fixes itself and I can make calls again. Any ideas on what is wrong? FWIW,
the phone line is not in use and I'm the only user.
Thanks,
Chris
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Re: [asterisk-users] DAHDI Congestion cause 34
On Sun, Apr 25, 2010 at 08:37:22PM +0300, Chris Datfung wrote:
> Hi,
>
> I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk and
> try to make a call I get the following error message:
>
>
> -- Executing [6781...@default:1] Dial("IAX2/iaxy-7477",
> "DAHDI/g1/96781948") in new stack
> [Apr 25 13:00:10] WARNING[3772]: app_dial.c:1806 dial_exec_full: Unable to
> create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
> == Everyone is busy/congested at this time (1:0/1/0)
> -- Auto fallthrough, channel 'IAX2/iaxy-7477' status is 'CONGESTION'
>
>
> Restarting DAHDI does not help, for somehow after several hours the problem
> fixes itself and I can make calls again. Any ideas on what is wrong? FWIW,
> the phone line is not in use and I'm the only user.
What device is that?
--
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
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Re: [asterisk-users] DAHDI Congestion cause 34
On Sun, Apr 25, 2010 at 8:43 PM, Tzafrir Cohen wrote:
> On Sun, Apr 25, 2010 at 08:37:22PM +0300, Chris Datfung wrote:
> > Hi,
> >
> > I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk
> and
> > try to make a call I get the following error message:
> >
> >
>
> > -- Executing [6781...@default:1] Dial("IAX2/iaxy-7477",
> > "DAHDI/g1/96781948") in new stack
> > [Apr 25 13:00:10] WARNING[3772]: app_dial.c:1806 dial_exec_full: Unable
> to
> > create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
> > == Everyone is busy/congested at this time (1:0/1/0)
> > -- Auto fallthrough, channel 'IAX2/iaxy-7477' status is 'CONGESTION'
> >
>
> >
> > Restarting DAHDI does not help, for somehow after several hours the
> problem
> > fixes itself and I can make calls again. Any ideas on what is wrong?
> FWIW,
> > the phone line is not in use and I'm the only user.
>
> What device is that?
>
Sorry about that.
lab:~# cat /proc/dahdi/1
Span 1: WCFXO/0 "Wildcard X101P Board 1" (MASTER)
1 WCFXO/0/0 FXSKS (In use) (SWEC: MG2)
Is there any other info that would be helpful to debug?
Thanks,
Chris
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[asterisk-users] hardware clock drift and CDR
Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. However, suppose I update system time at every hour and it sets +1 minute (due to a -1 minute drift). Suppose a call is dialed at 03:58 and lasts 4 "real" minutes. According to the updated system time, the call will have lasted 5 minutes (4+1 drift). How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for "call start" or also for "call end"? What Asterisk-related side-effects should I expect from a drifting clock? Thanks Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP gain
Hi, Are SIP gain parameters available in Asterisk 1.4/1.6? I'm wondering if I can increase transmission gain on SIP channels. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP monitoring tools
On Sat, 24 Apr 2010, Michael Wilson wrote: > I think DSL is 1/2 duplex and in most cases way to slow on the way Up for > VOIP. Not sure what country you're in, but DSL in the UK is full duplex and more than capable of handling VoIP, as I'm sure the millions of users who use it would attest. My own ADSL outgoing is 830Kb/sec and many of my customers enjoy 1.2Mb/sec outging. (With up to 24Mb incoming) Speed usually isn't the issue, and it's a trivial calculation that we all ought to be aware of to work out how many channels a given speed DSL line can handle.. The biggest issue I've had in the UK has been using shite quality ISPs. Get a good business quality ISP, then a good router which can do outbound traffic shaping and you're in with a fighting chance. The only half duplex connections I've used in recent years have been Wi-Fi - which looks like full duplex, but is really half duplex at the link layer. For the OP: Your biggest hurdle is going to be working out how your countrys DSL backhaul to the ISP works, and then which ISP to choose, or if it's ISP controlled all the way with not choice from the exchange, just how good they are and what sort of contention they impose on their DSL network. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware clock drift and CDR
On Sun, 25 Apr 2010, Vieri wrote: > Hi, > > I've noticed that one of my new servers (new mobo) if drifting slowly > backwards in time (in aprox. 24 hours, system time drifts back 5 > minutes). > > I have an ntpd process which is supposed to sync with a lan time server > but it's not quite working. So I'm launching a manual ntpdate or > ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. So I'd start by fixing ntpd. It really is the best way forward. > However, suppose I update system time at every hour and it sets +1 > minute (due to a -1 minute drift). Suppose a call is dialed at 03:58 and > lasts 4 "real" minutes. According to the updated system time, the call > will have lasted 5 minutes (4+1 drift). > > How does Asterisk CDR count the duration/billsec values? Does it rely on > system time ONLY for "call start" or also for "call end"? > > What Asterisk-related side-effects should I expect from a drifting > clock? Who cares. Just fix ntpd then your worys are gone. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CONNECTEDLINE(), progressinband=no and 183 before 180 (with latest trunk)
I don't expect my SIP provider to provide useful "Remote-Party-ID" information.
Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to
populate remote party information from a local database.
I am also using the "I" (upper case "i") option for Dial.
Generally I like to see to see the remote party name appear on the phone's
display as soon as a call is dialed... even if the the remote party is BUSY.
I ran into a problem where the Remote-Party-ID does not get displayed on the
caller's phone until the remote phone is answered.
I finally tracked this down to several things:
a) My SIP provider sends "183 Session Progress" and inband ringback prior to
sending a "180 Ringing".
b) The default sip.conf file that ships with asterisk suggests using
"progressinband=no" for polycom phones.
c) The "progessinband=no" setting prevents the "180 Ringing" from being
forwarded to the phone if it is received after the "183 Session Progress".
d) Called-Parity-ID appears to be only sent to the phone with "180 Ringing" and
"200 OK" responses.
# this the sequence of events that transpire:
-caller places call
-asterisk receives "183 Trying" from SIP provider and forwards it to the
caller's phone
-asterisk receives inband ringback from SIP provider and forwards it to the
phone (RTP)
-asterisk receives "180 Ringing" from SIP provider but does "not" forward it to
the phone.
-asterisk continues to receive more inband ringback from SIP provider and it
continues to forward it to the phone (RTP)
-remote party answers the phone
-asterisk receives "200 OK" from SIP provider; asterisk inserts
"Called-Party-ID" and then forwards it to the calling phone.
-the display on the caller's phone is finally updated; ringback stops and
someone at the other end says "hello".
There are two workarounds which will make the Called-Party-ID show up on the
phone before the call is answered:
i) Use "progressinband=never" even though the default sip.conf file recommends
against it.
The recommendation is presumably based on some old bugs in the Polycom phones
that no longer exists.
I am using recent Polycom firmware and did not notice any bugs.
Note however that the the display on the phone won't be updated if the remote
phone is "BUSY", which in my case is not ideal.
ii) Use the "r" option to "Dial". e.g. Dial(SIP/${ext...@x,300,Ir);
This has the advantage of updating the phone very quickly without waiting for
any respones from the SIP provider.
This may have side effects: ringback could hypothetically be produced when it
shouldn't be.
Questions:
Is there a reason why "Remote-Party-ID" is not sent to the phone as part of the
"183 Trying" message?
Could this be a configurable option?
Should the example sip.conf file continue to recommend "progressinband=no" for
Polycom phones?
-crjw--
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Re: [asterisk-users] Asterisk not recognizing ACK from an OK message?Help debuging SIP retransmit problem
On Sat, Apr 24, 2010 at 7:01 AM, David White wrote: > > call-id doesn't match? > > SIP/2.0 200 OK > ... > Call-ID: 2117388659-506...@82.158.83.xxx > ... > ACK sip:6615xx...@130.117.xxx.xxx SIP/2.0 > ... > Call-ID: 2117388659-506...@192.168.1.100 > ... > > I'm not sure, but I think that the part after the '@' must also match > throughout the dialog. A Grandstream bug? Thanks! It might be a bug, I'll contact Grandstream and post again if it turns out to be a bug. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does 'file' command work with asterisk genereted alaw file
Hi, I record an alaw file by asterisk's record monitor command, and i use linux's file command to check it information. file command recognized the alaw file as "DATA", is it correct? Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does 'file' command work with asterisk genereted alaw file
On Sunday 25 April 2010 23:22:21 Pham Quy wrote: > I record an alaw file by asterisk's record monitor command, and i use > linux's file command to check it information. > > file command recognized the alaw file as "DATA", is it correct? The file command works by recognizing certain header data in a file. As the alaw format has no header, that command won't recognize the data as any particular format, so it tells you "DATA" as its fallback. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Swaping out phones.
Hi Tony. Maybe you have already resolv this. I suppose the new phone is not registering so the peers table isn't updated. You may check if, when phone is turned on sends an registration request. Best Regards Jose Flores Galicia <> BriefCode && Code Based Training 2010/4/22 Tony LaMear > I have a quick question. I am using Asterisk 1.4. I have a user that has > changed phones (grandstream budge tone 200 to a polycom 330). I have changed > the sip.conf and extensions.conf. I have also unplugged the old phone and > plugged in the new phone. I get the ext showing on the phone, but when I do > a sip show peer 5000 the old ip address and phone show up. I did a sip > reload and a dialplan reload. Any other ideas? > > > > Thanks > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
