[asterisk-users] DONT_OPTIMISE, BETTER_BACKTRACES and performance
Hi How much impact on performance do DONT_OPTIMISE and BETTER_BACKTRACES have on a busy (13000+ entries in cdr for yesterday) server? I'm trying to decide whether to have them on in case of crashes or not. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Is it possible call land into extensions.ael configuration file not in extensions.conf
Hi List, I want to change the asterisk flow. right now call startd from extensions.conf. Is there any way by which we can changed it to extensions.ael or extensions.lua ? - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] Is it possible call land into extensions.ael configuration file not in extensions.conf
Hi, Create a context in AEL, or LUA and change the context=ael-context or context=lua-context in sip.conf [default] section or for each sip user decalred who needs to start call in context defined in AEL/LUA? Regards, Gohar From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 23, 2011 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind Subject: [asterisk-users] Is it possible call land into extensions.ael configuration file not in extensions.conf Hi List, I want to change the asterisk flow. right now call startd from extensions.conf. Is there any way by which we can changed it to extensions.ael or extensions.lua ? - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] MWI for non-subscribed Realtime peers?
Hi, I have an Asterisk behind an OpenSIPS proxy. The proxy handles registrations and also SIP SUBSCRIBE for MWI. The Asterisk are configured to send NOTIFY to the proxy even when the SUBSCRIBE haven't been received. I can configure a user in sip.conf that works: [az5134939706] type=friend host=xxx.xxx.xxx.xxx (IP of proxy) port=5060 nat=no mailbox=1234@customer subscribemwi=no defaultuser=az5134939706 Every time a voicemail has been left in the mailbox 1234@customer, a NOTIFY is sent off to the proxy. Remember, the peer doesn't register or send SUBSCRIBE to Asterisk, but subscribemwi=no forces NOTIFY to be sent anyway. However, I am struggling to get the same thing working for Realtime peers. I have rtcachefriends=yes set in sip.conf. But I never see the peer loaded from database and no NOTIFY is ever sent. Is it possible to user Realtime this way? What will trigger loading of the peer? Best regards, Jan Blom -- _ -- 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] Is it possible call land into extensions.ael configuration file not in extensions.conf
Hi Gohar, As per you suggestion I make context into AEL file and working file. But I do little bit RD on that case I make same context into both files(.conf and .ael) and asterisk read 1st .conf files extension. It means if we make anythings into AEL files then asterisk 1st check into .conf file then another one. It might be time consuming if we have Lot's off context. But any way thanks for you reply. On Wed, Nov 23, 2011 at 5:16 PM, Gohar Ahmed gohar.ah...@vopium.com wrote: Hi, Create a context in AEL, or LUA and change the context=ael-context or context=lua-context in sip.conf [default] section or for each sip user decalred who needs to start call in context defined in AEL/LUA? ** ** Regards, Gohar ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Wednesday, November 23, 2011 4:37 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind *Subject:* [asterisk-users] Is it possible call land into extensions.ael configuration file not in extensions.conf ** ** Hi List, I want to change the asterisk flow. right now call startd from extensions.conf. Is there any way by which we can changed it to extensions.ael or extensions.lua ? - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer ** ** -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] DONT_OPTIMISE, BETTER_BACKTRACES and performance
Am 23.11.11 11:39, schrieb Ishfaq Malik: Hi How much impact on performance do DONT_OPTIMISE and BETTER_BACKTRACES have on a busy (13000+ entries in cdr for yesterday) server? I'm trying to decide whether to have them on in case of crashes or not. Hi, IMHO a very big impact. for my system (50k calls per day) i had a load of average 5 on a dual 6 core machine. without DONT_OPTIMIZE the load of this server is around 0.5 ;) if you have to find a problem then use it and hope the best but you should avoid these settings if its not necessary. best regards stefan -- _ -- 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 use password file withAuthenticateApplication
hello, I wrote a post about this (spanish) http://www.voztovoice.org/?q=node/477 Or, if you prefere, using func_odbc http://www.voztovoice.org/?q=node/478 Regards - Bakko-- _ -- 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] safe_asterisk ?
Safe_asterisk refers to the bash script /usr/sbin/safe_asterisk which is installed by all Asterisk installs whether by rpm, tar or svn. It does exhibit daemon-like behavior in that it is run as a background process and will restart itself if you kill it incorrectly. From: virendra bhati [mailto:virbh...@gmail.com] Sent: Wednesday, November 23, 2011 1:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Sam Govind; Danny Nicholas Subject: safe_asterisk ? Hi List, What do it mean safe_asterisk ? I read too much about it but how it's works as Daemon process? When We install asterisk with the help of .tar file then safe_asterisk is install or not ? If yes then how can we work with that ? I am too much confusing.. - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] safe_asterisk ?
On Wed, Nov 23, 2011 at 08:16:36AM -0600, Danny Nicholas wrote: Safe_asterisk refers to the bash script /usr/sbin/safe_asterisk which is installed by all Asterisk installs whether by rpm, tar or svn. It does exhibit daemon-like behavior in that it is run as a background process and will restart itself if you kill it incorrectly. Note that systems with upstart / systemd will do that more relibly. If you run asterisk under one of those, use a plain upstart / systemd init config rather than a legacy sysv init.d script and avoid using safe_asterisk. -- 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 -- 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] MWI for non-subscribed Realtime peers?
Let me answer my own question. That may save someone's frustration in the
future.
The problem is that the Realtime peer never gets loaded, since SIP REGISTER and
SIP SUBSCRIBE never reaches Asterisk.
Doing a simple sip show peer az5134939706 load from CLI will force load of
peer. However, I needed a way of having this done automatically on startup for
all (many!) peers. A number of methods are suggested by people (use Google) but
they all seemed like hacks to me.
Finally I realized, after rereading chan_sip.c, the solution was to force load
the peer from dialplan. If I do this just before I send a caller to voicemail,
I can be sure the peer is available when MWI NOTIFY should be sent. Just add
this to the dialplan:
same = n,NoOp(${SIPPEER(az5134939706)})
Good luck with your Realtime MWI hacking!
Best regards,
Jan
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jan Blom
Sent: den 23 november 2011 13:04
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MWI for non-subscribed Realtime peers?
Hi,
I have an Asterisk behind an OpenSIPS proxy. The proxy handles registrations
and also SIP SUBSCRIBE for MWI. The Asterisk are configured to send NOTIFY to
the proxy even when the SUBSCRIBE haven't been received. I can configure a user
in sip.conf that works:
[az5134939706]
type=friend
host=xxx.xxx.xxx.xxx (IP of proxy)
port=5060
nat=no
mailbox=1234@customer
subscribemwi=no
defaultuser=az5134939706
Every time a voicemail has been left in the mailbox 1234@customer, a NOTIFY is
sent off to the proxy. Remember, the peer doesn't register or send SUBSCRIBE to
Asterisk, but subscribemwi=no forces NOTIFY to be sent anyway.
However, I am struggling to get the same thing working for Realtime peers. I
have rtcachefriends=yes set in sip.conf. But I never see the peer loaded from
database and no NOTIFY is ever sent.
Is it possible to user Realtime this way? What will trigger loading of the peer?
Best regards,
Jan Blom
--
_
-- 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] safe_asterisk ?
On 11-11-23 09:21 AM, Tzafrir Cohen wrote: On Wed, Nov 23, 2011 at 08:16:36AM -0600, Danny Nicholas wrote: Safe_asterisk refers to the bash script /usr/sbin/safe_asterisk which is installed by all Asterisk installs whether by rpm, tar or svn. It does exhibit daemon-like behavior in that it is run as a background process and will restart itself if you kill it incorrectly. Note that systems with upstart / systemd will do that more relibly. If you run asterisk under one of those, use a plain upstart / systemd init config rather than a legacy sysv init.d script and avoid using safe_asterisk. We should consider updating the Makefile in asterisk trunk to start using them. More and more OS are starting to support them. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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
Re: [asterisk-users] Follow me unreachable message default
1.4.8 is a pretty old version. The simplest option IMO would be to remove/replace the file in /var/lib/asterisk/sounds/followme. You can also check your followme.conf and features.conf settings. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier Sent: Tuesday, November 22, 2011 5:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Follow me unreachable message default Hey folks, been out of the loop for a while and need to make a few changes to my Ast box. I have been digging around trying to find an answer on the list archives, the wiki, google etc. but not joy. I have using Asterisk 1.4.8 and have recently added the FollowMe feature. Basically, I have what I need working but when nobody answers on any of the follow me numbers the caller hears the The party you are calling is unreachable message, then goes to my vmail. This is OK but I don't like the unreachable message and I want that skipped/removed from my config. I read in the docs that you are suppose to use the n option to enable this which I have NOT done anywhere in followme.conf or when I call the followme app from my dial plan. For some reason it's defaulting to this option. Is there a way to turn the playing of the unreachable message off, short of changing the message it plays to something more desirable? n - Playback the unreachable status message if we've run out of steps to reach the Thanks, Todd -- _ -- 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] safe_asterisk ?
This is the right place to ask the question: what is the best practice to run asterisk safe_asterisk-like method, but without _using_ safe_asterisk shell script. Actually, what is safe-asterisk doing? It restarts asterisk process and renames core dumps for better understanding of the scenario of crashing. I'm using stable debian, obviously there is no upstart, neither systemd (@ squeeze, of course), so for restarting asterisk i can use runit or something like this. How about this, guys, is somebody using runit (or similliar software) in production. For core dumps, i've read that you can tweak a little bit sysctl.conf kernel.core_uses_pid = 1 kernel.core_pattern = /var/tmp/core.%p.%e.%s fs.suid_dumpable = 1 So my question is: what are you guys using? 2011/11/23 Paul Belanger pabelan...@digium.com On 11-11-23 09:21 AM, Tzafrir Cohen wrote: On Wed, Nov 23, 2011 at 08:16:36AM -0600, Danny Nicholas wrote: Safe_asterisk refers to the bash script /usr/sbin/safe_asterisk which is installed by all Asterisk installs whether by rpm, tar or svn. It does exhibit daemon-like behavior in that it is run as a background process and will restart itself if you kill it incorrectly. Note that systems with upstart / systemd will do that more relibly. If you run asterisk under one of those, use a plain upstart / systemd init config rather than a legacy sysv init.d script and avoid using safe_asterisk. We should consider updating the Makefile in asterisk trunk to start using them. More and more OS are starting to support them. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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-**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] DONT_OPTIMISE, BETTER_BACKTRACES and performance
I think that you should use asterisk with this compile flags when you actually have deadlocks or often crashes. 2011/11/23 Stefan Schmidt s...@sil.at Am 23.11.11 11:39, schrieb Ishfaq Malik: Hi How much impact on performance do DONT_OPTIMISE and BETTER_BACKTRACES have on a busy (13000+ entries in cdr for yesterday) server? I'm trying to decide whether to have them on in case of crashes or not. Hi, IMHO a very big impact. for my system (50k calls per day) i had a load of average 5 on a dual 6 core machine. without DONT_OPTIMIZE the load of this server is around 0.5 ;) if you have to find a problem then use it and hope the best but you should avoid these settings if its not necessary. best regards stefan -- _ -- 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] Is it possible call land into extensions.ael configuration file not in extensions.conf
Sorry for the top post, this is from my phone. Asterisk parses all of the config files (.conf, .ael and .lua, assuming you have the appropriate modules loaded) at the time you load asterisk or reload the dialplan (dialplan reload). It does not read the files each time a new call is started. Thanks, --Warren Selby, dCAP On Nov 23, 2011, at 6:11 AM, virendra bhati virbh...@gmail.com wrote: Hi Gohar, As per you suggestion I make context into AEL file and working file. But I do little bit RD on that case I make same context into both files(.conf and .ael) and asterisk read 1st .conf files extension. It means if we make anythings into AEL files then asterisk 1st check into .conf file then another one. It might be time consuming if we have Lot's off context. But any way thanks for you reply. On Wed, Nov 23, 2011 at 5:16 PM, Gohar Ahmed gohar.ah...@vopium.com wrote: Hi, Create a context in AEL, or LUA and change the context=ael-context or context=lua-context in sip.conf [default] section or for each sip user decalred who needs to start call in context defined in AEL/LUA? Regards, Gohar From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 23, 2011 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind Subject: [asterisk-users] Is it possible call land into extensions.ael configuration file not in extensions.conf Hi List, I want to change the asterisk flow. right now call startd from extensions.conf. Is there any way by which we can changed it to extensions.ael or extensions.lua ? - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] How to use password file with AuthenticateApplication
On 11/22/11 9:02 PM, virendra bhati wrote: On Mon, Nov 21, 2011 at 6:15 AM, virendra bhati virbh...@gmail.com mailto:virbh...@gmail.com wrote: Hi, After deleting all space no improvements. Try reversing the account code and password hash, like this: 81dc9bdb52d04dc20036dbd8313ed055:Virendra 9996535e07258a7bbfd8b132435c5962:Vijay 7bccfde7714a1ebadf06c5f4cea752c1:VirendraBhati the original format (i.e. accountcode:md5hash) was correct. one question, when you create the md5hash did you use the echo command? if so, did you specify the -n option? e.g. echo -n 12345 | md5sum -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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] Question about Read() application
Danny,
Did you try my local channel idea?
Cheers,
Kingsley.
On Mon, 2011-11-21 at 08:25 -0600, Danny Nicholas wrote:
I tried to patch app_read on my development dahdi box as follows:
static int unload_module(void)
{
int res;
res = ast_unregister_application(app);
/* ast_module_user_hangup_all(); */
return res;
}
But the offending behavior persists - it's not a show-stopper but it
eventually could be.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart
Sent: Saturday, November 19, 2011 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about Read() application
Hi,
Did you get a workaround for this? I sent you a message offlist but you
didn't reply so I don't know whether you saw it.
Cheers,
Kingsley.
On Fri, 2011-11-18 at 13:15 -0600, Danny Nicholas wrote:
My IVR wouldn't sound right if I allowed 2 or 3 times before it was
considered a failure. The big(ger) problem is that it just hangs up when
it fails, no warning or work around to do.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug
Lytle
Sent: Friday, November 18, 2011 1:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about Read() application
Danny Nicholas wrote:
The user reported to me that I punched 1 and it hung up - in my
testing,
I found that slow DTMF entry (1 digit every 2 seconds or so) or fast
entry (more than 10 digits per second) was most likely to cause the
problem.
I've never had mine just hangup on a mis-key, but then again I have it
try 3 times before considering it a failure.
exten = s,1,Read(get-admin-password|enter-password|||3|)
Doug
--
_
-- 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
--
_
-- 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
