[asterisk-users] ulimit
Dear for having an stable system which limit option is good for ulimit comand ? 2-is any option for making asterisk crash-free? Best -- Pezhman Lali -- _ -- 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] BT killed Ribbit
Any thoughts on why they did this? - http://venturebeat.com/2011/08/09/bt-kills-ribbits-web-phone-platform-se nds-customers-to-the-fast-growing-twilio/ what makes Twillio successful but another company willing to kill off a $100m+ investment? Cheers, Dean -- _ -- 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] SRV question
Am 08.08.2011 18:37, schrieb J Gao: Hello, All, I have a question about using SRV record. One of SIP provider is using DNS SRV record. If I use IP address of the SIP proxy server I can successfully register my Asterisk 1.8.5. But If I try to use the domain name like: /register = user:p...@somedomain.com/ then the registration failed. So I have to /dig somedomain.com SRV/ to find out the SIP proxy host, for example, sip.regserver.com, then I have to run /dig sip.regserver.com/ to find out the real IP address. Does Asterisk has the feature so I can register by SRV record? I lookup up sip.conf and I enabled srvlookup=yes, but that is only for SIP URI outgoing call, not for SIP registration. Thank you in advance. Jian Did you set srvlookup = yes in your sip.conf? regards, Ruben -- _ -- 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] ulimit
is it possible to prevent 100% cpu usage by asterisk, with ulimit? On Wed, Aug 10, 2011 at 11:53 AM, Pezhman Lali l...@lopl.net wrote: Dear for having an stable system which limit option is good for ulimit comand ? 2-is any option for making asterisk crash-free? Best -- Pezhman Lali -- Pezhman Lali -- _ -- 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] block all numbers begin by 00 and 1
thanks A J stiles and thanks Marcelo for your help and support
i have remove 1 extension begin by _00 and _1 and all function without
issue :)
thanks and regards.
2011/8/9 Marcelo Ellmann Clemente ellm...@freeddom.com
That would work as well! :)
---
Marcelo Ellmann
Freeddom Tecnologia e Serviços S/A
+55 11 52133200 Ramal 1016
- Original Message -
From: A J Stiles asterisk_l...@earthshod.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 9 August, 2011 12:21:09 PM
Subject: Re: [asterisk-users] block all numbers begin by 00 and 1
On Tuesday 09 Aug 2011, salaheddine elharit wrote:
hello
i want to know how to do in order to block all numbers bgin by 00 and all
numebrs begin by 1
i use sip account
All you have to do is, just make sure that there is no extension in the
default context which matches _00. or _1. -- or, if there is one, it
contains
something other than Dial(${EXTEN}) .
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] ulimit
is it possible to prevent 100% cpu usage by asterisk, with ulimit? Hi, it is possible, but not recommended. There is a reason, why the asterisk process needs 100% of CPU. What is your scenario? How many extension? What type of extensions? (SIP, etc) How many concurrent calls? Hardware? You are trying to cure the symptom, not the cause ;-) regards, Ruben -- _ -- 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] Unable to enable echo cancellation on channel 1 (No such device)
Hi All; Suddenly, we restarted the Asterisk machine and the echo appeared. The lines are analoge. At the consol, I see this message: [Aug 10 14:36:05] WARNING[3789]: chan_dahdi.c:4831 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) [Aug 10 14:36:07] WARNING[3789]: chan_dahdi.c:4831 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) What could be the reason? The asterisk version is Asterisk 1.8.5.0 DAHDI Version: 2.4.1.2 Echo Canceller At the system.conf I have the following: echocanceller=mg2,1-16 fxoks=1-16 At the chan_dahdi.conf context=IncomingPSTN signalling=fxs_ks rxgain=0.0 txgain=0.0 channel = 1-16 group=1 channel = 1-16 The calls are working, but the echo is appearing. Something strange is the result of the dahd_cfg, but really if I changed the configuration in the system.conf and I made it fxs instead of fxo, then it is not working: [root@PBX-FF asterisk]# dahdi_cfg DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) Selected signaling not supported Possible causes: FXO signaling is being used on a FXO interface (use a FXS signaling variant) RBS signaling is being used on a E1 CCS span Signaling is being assigned to channel 16 of an E1 CAS span [root@PBX-FF asterisk]# Any help? Maybe the version I have has a bug? Or I have to do recompilation? Regards Bilal -- _ -- 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 1.8 Install Problem
Hi list, I have a problem with installing Asterisk (under https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages): sudo apt-get install asterisk-1.8 Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'asterisk' instead of 'asterisk-1.8' Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: asterisk : Depends: asterisk-core-sounds-en-gsm (= 1.4.21) but it is not going to be installed E: Broken packages Thank you for your help -- _ -- 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 1.8 Install Problem
On Wednesday 10 Aug 2011, A.H. Jos wrote: Hi list, I have a problem with installing Asterisk (under https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages): sudo apt-get install asterisk-1.8 Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'asterisk' instead of 'asterisk-1.8' Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: asterisk : Depends: asterisk-core-sounds-en-gsm (= 1.4.21) but it is not going to be installed E: Broken packages What distribution are you using? From the messages, it looks as though it could be Debian Sid or Wheezy, or possibly Ubuntu. You could try $ sudo apt-get install asterisk-core-sounds-en-gsm and see if that helps. What you might well have going on is a repository conflict, where apt wants to install incompatible packages from different repositories. I've personally never bothered with pre-compiled Asterisk packages. Just apt-get purge it (so that apt won't interfere with your installation in future), and build the latest version from the Source Code. -- AJS Answers come *after* questions. -- _ -- 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] fail to correctly build 1.8.5 ??
Am 08.08.2011 18:28, schrieb Kevin P. Fleming: On 08/08/2011 07:51 AM, Norbert Zawodsky wrote: Am 08.08.2011 13:38, schrieb Soeren Malchow (MCon): Do you see the loaded modules when not using the conf file, somethinglike this ? cdr_mysql.so MySQL CDR Backend 0 res_config_mysql.so MySQL RealTime Configuration Driver 0 app_mysql.so Simple Mysql Interface 0 soeren Hi Soeren, Yes, module show shows all three of them. Please check to see if there is an issue open for this problem on https://issues.asterisk.org/jira. If there is not, please open one; an incorrectly formatted configuration file should not result in a segfault. Hi Kevin, I searched JIRA but didn't find anything which sounds familiar. I will open an issue as soon as I'm sure that this problem isn't my fault. Why do you mean ...an incorrectly formatted conf file ...? I just copied cdr_mysql.conf.samle to /etc/asterisk/cdr_mysql.conf and enabled the line hostname=database.host.name by removing the comment-sign. I was a programmer in my previous life (but that was 20 years ago different OS). I'd like to track that down a bit but don't know how to start. Can you tell me how to compile with debug-info? Is there a commandline switch to use with configure? With make? Norbert -- _ -- 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 1.8 Install Problem
I am using Ubuntu 11.04. I had installed it properly the first time, after that I removed it (due to a problem with OpenBTS), Installed Asterisk from Source Code (problem with OpenBTS persists++), And when returned back to the Repository I have this problem!!! On Wed, Aug 10, 2011 at 9:07 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Wednesday 10 Aug 2011, A.H. Jos wrote: Hi list, I have a problem with installing Asterisk (under https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages): sudo apt-get install asterisk-1.8 Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'asterisk' instead of 'asterisk-1.8' Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: asterisk : Depends: asterisk-core-sounds-en-gsm (= 1.4.21) but it is not going to be installed E: Broken packages What distribution are you using? From the messages, it looks as though it could be Debian Sid or Wheezy, or possibly Ubuntu. You could try $ sudo apt-get install asterisk-core-sounds-en-gsm and see if that helps. What you might well have going on is a repository conflict, where apt wants to install incompatible packages from different repositories. I've personally never bothered with pre-compiled Asterisk packages. Just apt-get purge it (so that apt won't interfere with your installation in future), and build the latest version from the Source Code. -- AJS Answers come *after* questions. -- _ -- 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] SRV question
On 11-08-10 03:18 AM, Ruben Rögels wrote: Am 08.08.2011 18:37, schrieb J Gao: Hello, All, I have a question about using SRV record. One of SIP provider is using DNS SRV record. If I use IP address of the SIP proxy server I can successfully register my Asterisk 1.8.5. But If I try to use the domain name like: /register = user:p...@somedomain.com/ then the registration failed. So I have to /dig somedomain.com SRV/ to find out the SIP proxy host, for example, sip.regserver.com, then I have to run /dig sip.regserver.com/ to find out the real IP address. Does Asterisk has the feature so I can register by SRV record? I lookup up sip.conf and I enabled srvlookup=yes, but that is only for SIP URI outgoing call, not for SIP registration. Thank you in advance. Jian Did you set srvlookup = yes in your sip.conf? regards, Ruben -- _ -- 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 Yes I did, as I mentioned in my first email. And, again, that srvlookup is only for calling outbound using SIP URI dialing, not for SIP Registration, am I right? Jian -- -- _ -- 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] ulimit
Hi, I use: ulimit -c unlimited My asterisk box handles about 250 concurrent Channels Regards Juan. Linux User #441131 On Wed, Aug 10, 2011 at 2:23 AM, Pezhman Lali l...@lopl.net wrote: Dear for having an stable system which limit option is good for ulimit comand ? 2-is any option for making asterisk crash-free? Best -- Pezhman Lali -- _ -- 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] FAX Issues
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Lee Howard Sent: Tuesday, August 09, 2011 7:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Issues Ryan McGuire wrote: Unless your network is under load and you are seeing dropped packets and high jitter, I would absolutely not do T.38. The cheapest and easiest approach that I have found is to buy yourself an FXS gateway and just make sure you are using ulaw. As SIP is usually running over UDP/IP it doesn't take much to produce dropped packets. Dropped packets mean lost audio which means lost data and possible demodulation difficulties for the modems. If you're in an environment where dropped UDP packets don't occur you're in a very rare scenario. For the most part people who claim success when faxing over SIP G.711 are being rescued by ECM (error correction) within the fax protocol. There are very, very few who really have mitigated UDP packet loss. That said, all T.38 systems are not equal. Certainly, the reliability of your T.38 provider may not be any better than that of G.711 fax over the SIP UDP. I only recommend faxing over TDM everything else is at your own risk. Find a carrier, likely a CLEC, willing and able to quickly re-route numbers when you have an outage. This is telecom, you are going to have an outages caused by something totally out of your control. If you have a TN down for whatever reason, having a POTS fax machine or a couple of POTS lines into your PBX for your carrier to re-route the non-working TN to can save the day. Carriers use a variety of methods to give you dialtone on a pair of wires. Make sure the line is plain old boring analog all the way to your carrier's switch. Here is how I look at it. Assume 1 in 10 faxes fail when using ulaw with SIP on a nice stable QoS'd connection to your carrier . There is no specific percentage, read the mailing list archives; I don't think it is an outrageous assumption. Error Correction (ECM) can only do so much. For personal use, that might not be a big deal. For business use, it is likely to be a big deal. People get very annoyed when they don't get their faxes. If you don't need a large number of fax numbers and don't need to handle a large number of faxes, then use standalone POTS lines and fax machines. It is simple, reliable, and people are familiar with it. POTS is generally easy to troubleshoot, if the telco who provides the POTS service tells you it is an inside wiring issue or a fax machine issue, then chances are it is an inside wiring issue, PBX issue, handset issue, or fax machine issue. If you need a large number of fax numbers or need to handle a large number of faxes then you really should consider a PRI to back up your VoIP service -- use the PRI for large numbers of fax numbers pointing to app_fax, simple fax to PDF conversion scripts are on voip-info.org, e-mail it to a destination e-mail address. This is quite reliable, relatively easy to implement, uses well known, reasonably mature technology, and protocols. You can use something like NVFaxDetect or the built-in fax detection of DAHDI to do combined voice/fax telephone numbers. Enough people listen for the far end fax tone before pressing Send, iy can become an issue. TNs on PRIs are often very cheap compared to DIDs on VoIP, consider a dedicated fax TN for each person and avoid the hassle of fax detecting. There is a large community base and lots of documentation. If you are budgeting for a PRI then a POTS line is not going to be a large expense, might as well have one or two around when all else fails. T.38 is a fairly new protocol compared to POTS and PRI and even Asterisk and app_fax. The support community is much smaller, documentation is not as complete. You will likely need plenty of T.38 support from your carrier to get it working. If you need large numbers of fax numbers across a large area, or if you need fax numbers in places you have no presence , or any number of reasons, investigate T.38. Today T.38 seems to require a significant investment in research, trials, and failures compared to POTS or PRI fax. You need to decide if the advantages of T.38 are worth the investment in time. Others will have to comment on the option of Hylafax, IAXmodem, or BRI for fax. * The opinions above are my own. -- _ -- 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] SRV question
Dear you must set srv_lookup on in your config. some basic info about srv is here http://blog.lopl.net/?p=23 best On Tue, Aug 9, 2011 at 11:38 PM, J Gao j...@veecall.com wrote: ** Anyone? On 11-08-08 09:37 AM, J Gao wrote: Hello, All, I have a question about using SRV record. One of SIP provider is using DNS SRV record. If I use IP address of the SIP proxy server I can successfully register my Asterisk 1.8.5. But If I try to use the domain name like: *register = user:p...@somedomain.com* then the registration failed. So I have to *dig somedomain.com SRV* to find out the SIP proxy host, for example, sip.regserver.com, then I have to run *dig sip.regserver.com* to find out the real IP address. Does Asterisk has the feature so I can register by SRV record? I lookup up sip.conf and I enabled srvlookup=yes, but that is only for SIP URI outgoing call, not for SIP registration. Thank you in advance. Jian -- _ -- 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 -- Pezhman Lali -- _ -- 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] Increasing volume ?
On 08/03/2011 08:47 PM, Matt Riddell wrote: On 4/08/11 2:12 AM, Zeeshan Ali Shah wrote: Hi, I am running asterisk with konference . tried to increase the conference voice but not success i tried to add in diaplain SetGlobalVar(Set(VOLUME(TX)=10)) SetGlobalVar(Set(VOLUME(RX)=10)) Should be: SetGlobalVar(VOLUME(TX)=10) SetGlobalVar(VOLUME(RX)=10) Dialplan functions cannot be set globally. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] Asterisk+internal phones+recorded messages
Hi I want to change my old answering phone machine and two wireless phones with asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel 9133i) + Wifi/SIP phone I am wondering if I´ll lost actual functionalities that are present in my old answering machine: 1) is it possible to show the caller number (coming from PSTN/FXO) in both SIP phones (wifi/SIP and LAN phone) ? Does SIP protocol take in charge this functionality 2) Most important question is : can I see on those internal phones (Wifi/SIP phone and LAN phone) that I´ve some recoded messages on asterisk. Indeed, I have this fucntionality with my old answering machine where I can see the number of new messages recorded in a big LCD screen. Thx -- _ -- 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 reporting
Hallo, I have a production asterisk server running on Ubuntu however all my configs where done using the CLI. I would like to implement a reporting element into the server so I can know the number of calls made, for what duration, on what dates. What tool can I use that can fit within any already laid out dialplan? Thanks Richard Zulu Twitter www.twitter.com/richardzulu http://www.linkedin.com/in/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* On Thu, Aug 11, 2011 at 4:12 AM, neo haux neo.h...@gmx.com wrote: Hi I want to change my old answering phone machine and two wireless phones with asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel 9133i) + Wifi/SIP phone I am wondering if I´ll lost actual functionalities that are present in my old answering machine: 1) is it possible to show the caller number (coming from PSTN/FXO) in both SIP phones (wifi/SIP and LAN phone) ? Does SIP protocol take in charge this functionality 2) Most important question is : can I see on those internal phones (Wifi/SIP phone and LAN phone) that I´ve some recoded messages on asterisk. Indeed, I have this fucntionality with my old answering machine where I can see the number of new messages recorded in a big LCD screen. Thx -- _ -- 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
