Re: [asterisk-users] change channel variable to a user chosen value during a call
On Sat, 2012-09-01 at 10:05 +0200, Olle E. Johansson wrote: There is a hidden feature for SNOM phones in the SIP channel. They have a way to send a client code during the call (made for lawyers) and that will end up in the CDR. That is exactly what we needed Olle. Thanks! The use case is indeed for lawyers. Cheers, Frederic -- _ -- 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] iax2 trunks between asterisk servers
Stephen Collier wrote: Any ideas or suggestions appreciated. We keep an mysql database of all extensions (Fax2Email) that I use to do a lookup against the destination extension and then set the phone to display the name. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 Streaming MeetMe Conference
Hi, I managed to get this working on a brand new system, I suspect there was something strange going on on my test environment. Just using ices and a custom asterisk-ices.xml file did the trick. Adam - Original Message - From: Benny Amorsen benny+use...@amorsen.dk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, 16 September, 2012 12:35:58 PM Subject: Re: [asterisk-users] Asterisk Streaming MeetMe Conference Adam K. Dean a...@dmcip.com writes: Hi, I was wondering if anyone has any experience in streaming a MeetMe conference so that others might listen in to it? It would be nice if the audio format could be AAC, but at first any format will do. I did come across this: http://www.voip-info.org/wiki/index.php?page_id=991 Which looks interesting, but if anyone knows of a better way I would be interested! There's an example using Ices here: http://www.757.org/~joat/wiki/index.php?n=Main.HomebrewAsteriskConferenceManager Search for Streaming the conference. I'm not sure there is a better way that Ices; I think it's a pretty cool way. /Benny -- _ -- 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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
Took 3 days without errors like dahdi: Master changed to TE2/0/2 having installed the dahdi 2.6.1 But I have warnings that I copy below [Sep 17 09:25:27] WARNING[24233] mtp.c: MTP2 timer T3 timeout (failed to receive 'N', or 'E' after sending 'O'), initial alignment failed on link 'i1'. [Sep 17 09:25:28] NOTICE[24233] mtp.c: Got event on link 'i1': 5 (0/500). [Sep 17 09:25:28] NOTICE[2872] l4isup.c: Unhandled zaptel event 0x5 on CIC=30. [Sep 17 09:25:28] NOTICE[2872] l4isup.c: Unhandled zaptel event 0x4 on CIC=30. [Sep 17 09:25:28] NOTICE[24233] mtp.c: Got event on link 'i1': 4 (0/11). [Sep 17 09:25:29] NOTICE[2872] l4isup.c: Unhandled zaptel event 0x5 on CIC=30. [Sep 17 09:25:29] NOTICE[24233] mtp.c: Got event on link 'i1': 5 (0/500). [Sep 17 09:25:29] NOTICE[24233] l4isup.c: T1 timeout (waiting for RLC) CIC=20. [Sep 17 09:25:29] WARNING[24233] mtp.c: No signalling links inservice and no cluster receivers alive, dropping packet! [Sep 17 09:25:29] WARNING[24233] chan_ss7.c: MTP is now UP on link 'i1'. [Sep 17 09:25:29] NOTICE[24233] mtp.c: Sending TRA to peer on link 'i1' [Sep 17 09:25:29] WARNING[24233] mtp.c: Got SLTM with unexpected sls=1, OPC=13152 DPC=8458 on 'i1/16' sls=0, state=5. [Sep 17 09:25:30] NOTICE[24233] l4isup.c: Got GROUP RESET message, opc=0x3360, dpc=0x210a, sls=0x1, cic=17, range=14. [Sep 17 09:25:30] NOTICE[24233] l4isup.c: Got GROUP RESET message, opc=0x3360, dpc=0x210a, sls=0x1, cic=1, range=14. [Sep 17 09:25:58] NOTICE[24233] l4isup.c: Process CGU, cic=1, range=14 [Sep 17 09:25:58] NOTICE[24233] l4isup.c: Process CGU, cic=33, range=30 [Sep 17 09:26:00] NOTICE[24233] l4isup.c: Got GROUP RESET message, opc=0x3360, dpc=0x210a, sls=0x1, cic=33, range=30. [Sep 17 09:26:00] NOTICE[24233] l4isup.c: Process CGU, cic=17, range=14 [Sep 17 09:26:13] NOTICE[2925] l4isup.c: Short read on linkset siuc CIC=24 (read only 0 of 160) errno=11 (Resource temporarily unavailable) (supressed 0). [Sep 17 09:26:58] NOTICE[24233] l4isup.c: Process CGU, cic=1, range=14 [Sep 17 09:27:00] NOTICE[24233] l4isup.c: Process CGU, cic=17, range=14 [Sep 17 09:27:17] NOTICE[24233] mtp.c: Got status indication 'OS' while INSERVICE on link 'i1'. [Sep 17 09:27:17] WARNING[24233] chan_ss7.c: MTP is now DOWN on link 'i1'. [Sep 17 09:27:17] NOTICE[24233] mtp.c: MTP changeover last_ack=34, last_sent=34, from schannel 16, no INSERVICE schannel found [Sep 17 09:27:17] NOTICE[24233] mtp.c: Failover not possible, no other signalling link and no other host available. [Sep 17 09:27:17] WARNING[24233] chan_ss7.c: MTP is now DOWN on link 'i1'. [Sep 17 09:27:17] NOTICE[2947] l4isup.c: Unhandled zaptel event 0x4 on CIC=2. [Sep 17 09:27:17] NOTICE[24233] mtp.c: Got event on link 'i1': 4 (0/500). [Sep 17 09:27:19] NOTICE[24233] mtp.c: Got event on link 'i1': 5 (0/500). [Sep 17 09:27:19] NOTICE[2947] l4isup.c: Unhandled zaptel event 0x5 on CIC=2. [Sep 17 09:27:19] NOTICE[2947] l4isup.c: Unhandled zaptel event 0x4 on CIC=2. [Sep 17 09:27:19] NOTICE[24233] mtp.c: Got event on link 'i1': 4 (0/11). [Sep 17 09:27:19] WARNING[24233] mtp.c: MTP2 timer T3 timeout (failed to receive 'N', or 'E' after sending 'O'), initial alignment failed on link 'i1'. [Sep 17 09:27:21] NOTICE[24233] mtp.c: Got event on link 'i1': 5 (0/500). [Sep 17 09:27:21] NOTICE[2947] l4isup.c: Unhandled zaptel event 0x5 on CIC=2. [Sep 17 09:27:21] WARNING[24233] chan_ss7.c: MTP is now UP on link 'i1'. [Sep 17 09:27:21] NOTICE[24233] mtp.c: Sending TRA to peer on link 'i1' [Sep 17 09:27:21] WARNING[24233] mtp.c: Got SLTM with unexpected sls=1, OPC=13152 DPC=8458 on 'i1/16' sls=0, state=5. [Sep 17 09:27:22] NOTICE[24233] l4isup.c: Got GROUP RESET message, opc=0x3360, dpc=0x210a, sls=0x1, cic=17, range=14. [Sep 17 09:27:22] NOTICE[24233] l4isup.c: Got GROUP RESET message, opc=0x3360, dpc=0x210a, sls=0x1, cic=1, range=14. On Fri, Sep 14, 2012 at 1:09 PM, Shaun Ruffell sruff...@digium.com wrote: [friendly request that you inline or bottom post and trim your replies. See http://brooksreview.net/2011/01/interleaved-email/ which makes most of the points I would make on the subject] On Fri, Sep 14, 2012 at 09:00:39AM -0300, equis software wrote: My test were... SIEMENS - LTG1 --- cable1 - SS7 --- IVR1 (have errors) |- LTG2 --- cable2 - SS7 --- IVR2 (OK) minutes later... SIEMENS - LTG1 --- cable1 - SS7 --- IVR2 (OK) |- LTG2 --- cable2 - SS7 --- IVR1 (have same errors) And your problem follows the server not the card or the cable? Have you tried other slots? Are you screwing the cards down? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
On Mon, Sep 17, 2012 at 09:33:12AM -0300, equis software wrote: Took 3 days without errors like dahdi: Master changed to TE2/0/2 having installed the dahdi 2.6.1 A heads up: With DAHDI-Linux 2.5.0 and up you will only see the Master changed messages when you set the debug module parameter [1] [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=9299 You can enable this flag without reloading like: $ echo 1 /sys/module/dahdi/parameters/debug Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
[asterisk-users] AGI problem
Hello All, I need to use agi to handle some issue , after finishing agi i want to hang up the channel , if i call from an extension there is no problem but i want to be the same for PSTN (outside) caller , if someone call asterisk show the hang up channel but the caller is not disconnected and if meanwhile someone inside try to call from an extension the outide caller can listen to DTMF and everything . . . . I would be really grateful if you share your close experience . Regards, Mehdi -- _ -- 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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
and I was happy without this message!!! Thanks Shaun On Mon, Sep 17, 2012 at 11:22 AM, Shaun Ruffell sruff...@digium.com wrote: On Mon, Sep 17, 2012 at 09:33:12AM -0300, equis software wrote: Took 3 days without errors like dahdi: Master changed to TE2/0/2 having installed the dahdi 2.6.1 A heads up: With DAHDI-Linux 2.5.0 and up you will only see the Master changed messages when you set the debug module parameter [1] [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=9299 You can enable this flag without reloading like: $ echo 1 /sys/module/dahdi/parameters/debug Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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 -- _ -- 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] AGI HANGUP PROBLEM
Hi all,
I need to handle a problem from AGI please guide me
in extensions_custom.conf :
exten = s,1,Answer
exten = s,n,AGI(hang.php)
exten = s,n,Hangup
in hang.php :
#!/usr/bin/php -q
?
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$agi-answer();
$agi-say_number('1');
$agi-hangup();
?
calling from an extension has no problem but whenever i use landline
or mobile it can not hangup the call and the caller has to hangup the
call.
if the caller does not hangup the call it becomes kind of SPY (the
caller can listen DTMF if someone call from an extension)
I am using elastix 2.3.0 which has asterisk 1.8.10.0 .
I really appreciate your sharing.
Regards,
Mehdi
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[asterisk-users] inboun routing based on area aode
I am currently using AsteriskNow v2.
What I am trying to accomplish is having all calls from an area code go
directly to the person responsible for that area. While searching for a
solution for this I did come across a post that had a few examples. So Josh at
extension 1902 would receive all calls from the 808 area code.
exten = s,1,GotoIf($${CALLERIDNUM:0:3} = 808?1902|1)
While asterisknow uses freepbx to control the config files. Where and how would
I go about putting this into freepbx or another loaded config file that where
something like the above would work. Thanks,
/Josh
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Re: [asterisk-users] inboun routing based on area aode
On Mon, Sep 17, 2012 at 2:39 PM, Josh Hopkins j...@prorivertech.com wrote:
I am currently using AsteriskNow v2.
What I am trying to accomplish is having all calls from an area code go
directly to the person responsible for that area. While searching for a
solution for this I did come across a post that had a few examples. So
Josh at extension 1902 would receive all calls from the 808 area code.
exten = s,1,GotoIf($${CALLERIDNUM:0:3} = 808?1902|1)
That's valid, and we use this methodology:
exten = (pattern match destination)/_800NXX,1,Voicemailetc
While asterisknow uses freepbx to control the config files. Where and how
would I go about putting this into freepbx or another loaded config file
that where something like the above would work. Thanks,
You will usually get better answers for those products by going to lists
and forums dedicated to them. I don't think anyone here uses those.
--
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] inboun routing based on area aode
On 9/17/2012 4:39 PM, Josh Hopkins wrote: snip While asterisknow uses freepbx to control the config files. Where and how would I go about putting this into freepbx or another loaded config file that where something like the above would work. Thanks, /Josh Josh, You may want to look into using extensions_custom.conf along with Custom Extensions, Custom Destinations, Miscellaneous Applications, and Miscellaneous Destinations. Regards, Vladimir -- _ -- 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 on VM with NO DAHDI hardware
Thanks Shaun. On Sep 15, 2012 1:10 AM, Shaun Ruffell sruff...@digium.com wrote: On Fri, Sep 14, 2012 at 09:42:23PM -0400, Mark Robinson wrote: I did some research on this subject and still do not understand. Why we use modules if asterisk can obtain timing directly from kernel? Probably longer than it needs to be but... Asterisk needs timing information via a kernel file descriptor that can be passed into to the poll() [1] or select() system calls. This allows Asterisk to sleep while waiting for either a network packet to come in or one of several time intervals to expire in the same system call which is very efficient. DAHDI historically provided this service via /dev/dahdi/timer. DAHDI was a good choice to provide a timing service since Asterisk typically wanted VOIP traffic synchronized with any telephony hardware installed. If there was not telephony hardware DAHDI could use internal kernel interfaces to find the best source of timing for Asterisk (generic kernel timers, Real-Time Clock, High Resolution timers, etc..). Using DAHDI was (and is) a natural choice but updating kernels can be more difficult since DAHDI must be recompiled every time the kernel is updated. This unnecessarily increases the administration burden when there is not any telephony hardware to synchronize with. To reduce the dependency on DAHDI, res_timing_pthread was created in Asterisk version 1.6.1 [2]. This implementation could provide timing via file descriptors without requiring DAHDI to be installed. It uses a pipe()[3] and a thread. The thread will sleep for a period of time and then write to one end of the pipe when the timer fires. res_timing_pthread is more system intensive--creates a thread that must be scheduled, the scheduling of that thread determines the timeliness of the timer firing, and the extra system calls required to accomplish the same task--and I believe it is advisable to avoid res_timing_pthread unless you have no other choice. It was not until kernel version 2.6.25 that Linux provided a standard interface for configuring timers that signaled via file descriptors, the timerfd interface [4]. Now Asterisk had a standard kernel interface which provided essentially the same service that DAHDI's /dev/dahdi/timer did without needing to install DAHDI or creating a separate thread which wrote to a pipe and res_timing_timerfd was first release in Asterisk 1.6.2 [5]. The problem is that timerfd_create() is not available on all platforms Asterisk must support. Also, if you have telephony hardware installed it is still generally best to synchronize to the clock on the telephony hardware to minimize audio problems caused by mismatches in clock rates. But if available and you don't have any other dependency on DAHDI (app_meetme, app_page, etc...), timerfd should be your first choice for timing source. So given these different methods of obtaining timing, Asterisk needed a way to abstract them so that other parts of the system had a standard way to get a file descriptor which could be configured to fire at certain intervals. That is why the timing interface [6] and various timing modules were created. It allows the Asterisk administrator to use the timing source that works best for them. That's why, to the best of my knowledge, Asterisk uses modules even though it can now obtain timing directly from the kernel. Cheers, Shaun [1] http://linux.die.net/man/2/poll [2] http://svnview.digium.com/svn/asterisk?view=revisionrevision=122928 [3] http://linux.die.net/man/2/pipe [4] http://linux.die.net/man/2/timerfd_create [5] http://svnview.digium.com/svn/asterisk?view=revisionrevision=157820 [6] http://svnview.digium.com/svn/asterisk?view=revisionrevision=122062 -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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 -- _ -- 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] Need to record user voice while play background music
Hi Steve and Friends, Thanks for suggestion and help. I have done by using Monitor() for voice recording while play background music. I know other option is also possible to use MixMonitor(). -- Best Regards, Rajnikant Vanza Consultant Technology --- Working On Linux,C/C++,VoIP Technology On Fri, Sep 14, 2012 at 8:40 PM, Steve Edwards asterisk@sedwards.comwrote: Un-top-posting... On Fri, 14 Sep 2012, RAJNI VANZA wrote: I was wondering if anyone has any experience for recording user voice while play background music? On Fri, Sep 14, 2012 at 11:13 AM, Steve Edwards wrote: What methods have you tried? On Fri, 14 Sep 2012, RAJNI VANZA wrote: I have tried with Monitor(), MixMonitor() and conference (meetme) in dilaplan. By using Monitor() in dialplan recording is done through in, out two file mix in one recording file created but its not so good result. I'd vote for mixmonitor(). I use it to record calls in a 'third party verification' system. Interestingly (at least in 1.2), you can record the entire call in 1 file (for debugging) while recording parts in other files for delivery to the client. The 'parts' are assembled using sox in an AGI at the completion of the call. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- 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-**usershttp://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] iax2 trunks between asterisk servers
Doug, Thanks, that answers my question I will reuse the macro I'm using with an Avaya connection and CONNECTEDLINE(). Pity I was hoping iax2 would transfer callee id. Cheers Stephen -- _ -- 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 Test Suite error
Hi Matthew , i have enabled the framework and tested the script, after running i am getting some FAILS - tests/channels/SIP/refer_replaces_to_self --- FAILED -- tests/channels/SIP/sip_tls_call --- FAILED -- tests/channels/SIP/sip_cause --- FAILED -- tests/masquerade --- FAILED let me know still what i am missing in the testsuite. Regards Upendra On Thu, Sep 6, 2012 at 6:41 PM, Matthew Jordan mjor...@digium.com wrote: - Original Message - From: upendra uppi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, September 6, 2012 2:22:24 AM Subject: [asterisk-users] Asterisk Test Suite error Hi, i am trying to install the Asterisk test suite on my ubuntu system , i have followed all the installlation steps as mentioned in the link ( http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/ ) , but when i am trying to run the script some of the test cases are PASSED and most of them are FAILED and SKIPPPED. So please help me out to do the testing correctly. The following is the test report i got after running the script. snip There is newer documentation on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation There are pages that describe how to install and run the Test Suite under that page. Skipped tests happen when you are missing a dependency that the test needs to run. If you look at a verbose report for each test (which should also be output when the test run finishes), it should tell you what dependency was missing. Most of your failed tests appear to be those that require the TEST_FRAMEWORK compile time option in Asterisk. You can enable that in menuselect when you have configured Asterisk with --enable-dev-mode. Those tests *should* have picked up the fact that the TEST_FRAMEWORK wasn't enabled and should have been skipped (so long as the Asterisk directory that the Test Suite is sitting in has that option enabled in its build options), but there is a known bug with those test's YAML configuration that is preventing them from picking up the TEST_FRAMEWORK flag. Until we can get that cleaned up, enabling the TEST_FRAMEWORK flag should resolve that problem. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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 -- _ -- 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
