Re: [asterisk-users] Virtual machine timing (KVM)
David Backeberg wrote: On Fri, Feb 19, 2010 at 11:42 AM, Mike A. Leonetti mleone...@evolutionce.com wrote: To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? You could always use ConfBridge(), starting in 1.6.2.*, which does not require DAHDI/Zaptel, and therefore doesn't require a timer. Let me be the first to tell you that using a virt for a conferencing solution, especially if you want people to actually use it, sounds like a 'Bad Idea'. You could oversubscribe the resources so you don't starve the virt, but we already have a name or that. It's called not using a virt in the first place. Well, when you're right you're right. If it's really that much of a bad idea I'll just put in for a real machine. Although virtualizing seems to be all the buzz lately so I was just wondering if I could consolidate hardware (or continue to consolidate hardware). Our internal Asterisk does currently run on KVM. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual machine timing (KVM)
Sean Brady wrote: To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? The 2.2.1 version of DAHDI using DAHDI dummy seems to be working adequately in a Xen environment on CentOS for me, although I haven't been using MeetMe. Have you run into issues with it specifically? Which version of DAHDI are you using? If there are some issues that you have found I would like to know... Thanks, Sean To be honest I haven't tried it with Asterisk version 1.4 or higher. I only tried it with 1.2 and when the DAHDI was called Zaptel. I have been a little afraid to upgrade to 1.6 from 1.2 just in case there are some incompatibilities in my config that'll bring down the phone system here at the office for a while. The issue that I had was that the even the calls were choppy. Not even specifically just the MeetMe ones. But that was on VirtualBox. I am using KVM now. I'm not sure if that matters. What is your timer frequency set to in the kernel btw? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT: Has SILK codec gotten anywhere?
On 02/21/2010 02:02 AM, Kyle Kienapfel wrote: Hi, I stumbled upon mentions of a SILK codec last night on skypes skype for sip information page. I tried looking into it further and found some blog and mailing list posts from 2009 but I can't find any mentions of anything other than skype using the codec. Has the codec not gotten anywhere so far? http://en.wikipedia.org/wiki/SILK Skype took an extremely long time to start getting usable code into people's hands. It is starting to happen now, though. There seem to be several projects with support in the pipeline. Others may find the funky licencing a stumbling block. If the IETF codec works makes good progress its possible SILK will open up and be more widely usable. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual machine timing (KVM)
On Sun, Feb 21, 2010 at 06:56:04AM -0500, Mike A. Leonetti wrote: Well, when you're right you're right. If it's really that much of a bad idea I'll just put in for a real machine. Although virtualizing seems to be all the buzz lately so I was just wondering if I could consolidate hardware (or continue to consolidate hardware). Our internal Asterisk does currently run on KVM. DAHDI 2.2.1 works nicely with both internal timing and host DAHDI hardware. At least on my laptop. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] add Reason header on hangup
Hello, I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup: Reason: q.850;cause=17 Thanks -- Best Regards, Giedrius -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax, T38 and NAT
Magnus Benngård wrote: Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38 on 0197673581, changing t38pt_udptl=no to t38pt_udptl=yes,fec and try to send from 0197673581 to 0851711201, it is not working, switches to T38 sendimg a lot of UDPTL packages but it looks like (at least for me) that addresses are wrong. UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) 90.230.92.67 is WAN ip of 0197673581's router. 10.242.20.149 is ip of 0851711201's ATA (SPA2102). Shouldn't the UDPTL stream go through Asterisk? Have i missed sometheng else? Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) 0197673581 disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) 0851711201 disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=yes context=inputinterior.se directmedia=yes dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=no qualify=yes sendrpid=pai t38pt_udptl=yes,fec transport=udp trustrpid=yes type=friend videosupport=no [0851711290] secret=xyz callerid=Input Interior Sundbyberg (fax) 0851711290 ... rest is the same as [0851711201] Regards, Magnus Maybe you should give t38pt_usertpsource=yes a try. Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] add Reason header on hangup
What I do is have this:
exten = h,1,UserEvent(DialHungUp,ActionID:${CfMC_ActionID} ${UNIQUEID}
${CHANNEL} ${CfMC_AgentToUse} ${CfMC_DialInfo} ${CfMC_QueueToUse}
${HANGUPCAUSE} ${DIALSTATUS})
And then in AMI I know hangupcause and dialstatus.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Feb 21, 2010, at 7:14 AM, voipas wrote:
Hello,
I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup:
Reason: q.850;cause=17
Thanks
--
Best Regards,
Giedrius
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax, T38 and NAT
t38pt_usertpsource=yes seems to do the trick, switches to T38 and fax seems to go through (cant be 100% sure, the fax i am sending to is 500 km avay from me, but i dont get any errors and my fax thinks everything is ok, so I cross my fingers),,, On Sun, 21 Feb 2010 16:36:42 +0100, Johann Steinwendtner wrote: Magnus Benngård wrote: Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38 on 0197673581, changing t38pt_udptl=no to t38pt_udptl=yes,fec and try to send from 0197673581 to 0851711201, it is not working, switches to T38 sendimg a lot of UDPTL packages but it looks like (at least for me) that addresses are wrong. UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) 90.230.92.67 is WAN ip of 0197673581's router. 10.242.20.149 is ip of 0851711201's ATA (SPA2102). Shouldn't the UDPTL stream go through Asterisk? Have i missed sometheng else? Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=yes context=inputinterior.se directmedia=yes dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=no qualify=yes sendrpid=pai t38pt_udptl=yes,fec transport=udp trustrpid=yes type=friend videosupport=no [0851711290] secret=xyz callerid=Input Interior Sundbyberg (fax) ... rest is the same as [0851711201] Regards, Magnus Maybe you should give t38pt_usertpsource=yes a try. Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] add Reason header on hangup
21 feb 2010 kl. 16.14 skrev voipas: Hello, I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup: Reason: q.850;cause=17 No, you will have to change the code. I think there's a patch in the bug tracker. Go search on issues.asterisk.org. We do add a similar header if you check the BYE headers. X-Asterisk-hangup-cause. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi Congestion status
Hi, I'm using an half T1 line on a asterisk (obviously :)) 1.6.2.4 system, up to recently everything was fine but we are starting to experience the call limitation of the line (15). So as to warn user of the problem i attached a vocal notification to the CONGESTION status after a Dial(), but it looks like it also catch other congestion case (maybe on the receiver side). Should i / Could i use the ChanIsAvail() on a Dahdi channel ? will it work better to detect a full group ? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble with externalIVR socket connection
I'm having trouble with ExternalIVR's socket connection. Is it working in the current 1.6.0 trunk? I'm getting this error: Executing [...@ck987_externivr:1] ExternalIVR(SIP/itp_jnctn-006c, ivr://127.0.0.1:9012) in new stack [Feb 21 13:38:12] NOTICE[15864]: app_externalivr.c:631 eivr_comm: SIP/itp_jnctn-006c: stderr: Failed to execute 'ivr://127.0.0.1:9012': No such file or directory [Feb 21 13:38:12] WARNING[15864]: app_externalivr.c:625 eivr_comm: SIP/itp_jnctn-006c: Child process went away == Spawn extension (ck987_externIVR, s, 1) exited non-zero on 'SIP/itp_jnctn-006c' Here's my Dialplan: [ck987_externIVR] exten = s,1,ExternalIVR(ivr://127.0.0.1:9012) exten = s,n,Hangup() I'm running netcat in listen mode in a separate shell: [ck...@asterisk ~]$ nc -l 9012 * I've tried the standard ivr port (2949) as well. I was using 9012 because I've used that port for AGI socket connections on this box so I'm pretty sure that port is open. * I've tried using localhost instead of 127.0.0.1 * I've tried adding n as an option, just in case there had to be an option. * I'm running Asterisk 1.6.0.20 on Linux Red Hat 4.1.2, x86_64 Before I submit a bug I want to make sure I'm not missing something. Thanks, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HFC-S card
Does any one put a HFC-S card working in nt ptp mode? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: Dahdi Congestion status
Yes, it will catch congestion cases on the receiver side. There's two ways to avoid it: 1) use ChanIsAvail on the dadhi channel. It works very well. Or 2) Create a special signal for dahdi congestion (require modifying the source and recompile) that differs from the congestion cause. Here we created code 130, so we know when the congestion is on the dhadi channels or if it is an external/operator congestion case. We have servers running both ways, even with combination between ChanisAvail() and the special 130 code. All works very well. PS: ChanisAvail may fail sometimes if you have few available channels and a high demand for incoming calls. Exemple: suppose you have just one channel available, ChanisAvail will return this channel, but before dialplan reaches the Dial() command, an incoming call is received in this channel. So you don't have a channel do dial anymore, but ChanisAvail told you had one. You will then receive a congestion with code 34. Prado -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Benoit Enviada em: domingo, 21 de fevereiro de 2010 2:22 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] Dahdi Congestion status Hi, I'm using an half T1 line on a asterisk (obviously :)) 1.6.2.4 system, up to recently everything was fine but we are starting to experience the call limitation of the line (15). So as to warn user of the problem i attached a vocal notification to the CONGESTION status after a Dial(), but it looks like it also catch other congestion case (maybe on the receiver side). Should i / Could i use the ChanIsAvail() on a Dahdi channel ? will it work better to detect a full group ? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime extensions
On 19/02/10 8:15 AM, jonas kellens wrote: How about something like : [mycontext] exten = 100,1,NoOp(calling 100) exten = 100,n,NoOp(going realtime) switch = Realtime/mycont...@realtime_extensions mailto:mycont...@realtime_extensions ; from here on we use realtime And then my MySQL-DB contains : `extensions_table` VALUES (1, 'mycontext', '100', n, 'Wait', '2'); `extensions_table` VALUES (2, 'mycontext', '100', n, 'NoOp', 'into RealTime'); 'extensions_table` VALUES (3, 'mycontext', '100', n, 'Playback', 'my-sound-file'); I'm not sure that's likely to work - or if it does, not in the way you expect. Likely if you did a query for exten = 100, the n extensions would be returned in a random order. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime extensions
On 20/02/10 10:53 PM, jonas kellens wrote: I have read on this list that people do not get a reply if they ask stupid questions. Is this then a stupid question that I ask ? If nobody has ever combined extensions.conf and realtime in a way that I want to do, I wanna hear it too. Even if this means no solution for me. Then I know it's not doable. :) Maybe you should read the messages from the list then :) You've already been replied to. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Free iPhone Asterisk Function and Application Reference
Hi all, I've uploaded a free app for the iPhone called AsteriskRef to the Apple AppStore. This allows you to lookup applications and functions using your iPhone or iPod touch so you don't have to jump out of extensions.conf or open another terminal tab. It currently supports applications and functions from Asterisk 1.4, but I'm adding 1.6 and trunk at the moment. It currently requires OS3.1.3, but I've got another version under review at the moment which will run on 3.0. Hope you like it, let me know if you have any questions. More info here: http://www.venturevoip.com/news.php?rssid=2353 -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual machine timing (KVM)
Sean Brady wrote: To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? The 2.2.1 version of DAHDI using DAHDI dummy seems to be working adequately in a Xen environment on CentOS for me, although I haven't been using MeetMe. Have you run into issues with it specifically? Which version of DAHDI are you using? If there are some issues that you have found I would like to know... Thanks, Sean To be honest I haven't tried it with Asterisk version 1.4 or higher. I only tried it with 1.2 and when the DAHDI was called Zaptel. I have been a little afraid to upgrade to 1.6 from 1.2 just in case there are some incompatibilities in my config that'll bring down the phone system here at the office for a while. The issue that I had was that the even the calls were choppy. Not even specifically just the MeetMe ones. But that was on VirtualBox. I am using KVM now. I'm not sure if that matters. What is your timer frequency set to in the kernel btw? With DAHDI dummy in 2.2.1 you don't have to even do that, AFAIK. At least I didn't on my test box. I do get choppy audio when playing recordings occasionally. I haven't had time to figure that one out, but I haven't put it into production yet. I have been told repeatedly that Asterisk shouldn't be virtualized, and that timing was an issue, however I have never been given a reason that I consider acceptable to preclude me from doing so. I have also seen presentations talking about using Asterisk in Xen environments as well as Amazon's EC2 (also Xen). So there is some real contradictions and FUD surrounding Asterisk virtualized. Perhaps I am just stubborn, but I am determined to run Asterisk virtualized in production with conferencing (be it meetme or confbridge) until it's been proven without doubt that it just doesn't work. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio to remote AGI server
Several people in the past month have asked about sending audio of the call to
a remote AGI server. Currently, this is not available, because when we
connect to a remote AGI server, we connect on a single socket, which
establishes a command channel. I've been thinking about how we might
accomplish a secondary audio channel, and I've come up with what I believe
is a good solution. I've created a patch which establishes the following
commands:
open audio {tcp|udp} hostname portno
close audio
These commands would specify to what location audio data from the call would
be sent. Due to the feature policy of Asterisk, these would not become a part
of Asterisk until 1.8. However, because AGI commands are expandable at load
time, these commands could very easily be added to 1.4 and 1.6.x, via a third
party module (which I am considering writing, once this feature is in trunk).
What I need from the user community is for those who are interested to take a
look and possibly test this new patch. The patch is currently in design
phase, but testing could begin fairly soon. If you're interested in this
functionality, I invite you to take a look and comment on the proposed syntax,
as well as how the proposed usage would mesh with your individual needs.
https://issues.asterisk.org/view.php?id=16879
--
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 --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Playback will answer the call?
hi, all in my test,it shows Playback will answer the call automaticly, but i don't want to so. i will use answer function to answer the call. could you help me ? -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HFC-S card
On Sun, Feb 21, 2010 at 07:55:39PM +, Pedro Santos wrote: Does any one put a HFC-S card working in nt ptp mode? Which version of Asterisk do you use? Which channel driver? -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
