similar to: q: am i mixing somethign up?

Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc

G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO

Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . Regards H.Motamedi --------------

I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. -------------- next

Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: max_connection=1000 wait_timeout=60 query_cache_type=1 query_cache_limit=4M query_cache_size=512M interactive_timeout=120

Hi, My Dial() is set to the following, but always stops about 30 seconds into the call even when I set it to try for 60 seconds. exten => dialnumber,1,Dial(${DialInfo},60) I am running on 1.6.1-r199820. Is there some other setting that is overriding mine? Or an issue with this release? Thanks for the help. JR -------------- next part -------------- An HTML attachment was

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm

Hi, I was looking round on the Internet and saw there was no definitive list of free applications available for use with Asterisk, so I thought I'd compile a list for you all. If there's anything that you know of that is actively maintained but not in the list below, let me know (bear in mind I'm not including distros or Asterisk packagings in this list). Hopefully there are a few

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus

Hello Anyone who have already use/configure Asterisk with CDRTool ? Or maybe can suggest another CDR GUI ? regards. Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090812/e3e9e675/attachment.htm

Dear All Can you please do me favor and let me know what is my problem with my Asterisk VoiceMail configuration as it doesn't work correctly in my case ? Please find below that part of my extensions.conf that I intend to make use of voice mail for No Answer reply : " [line-incoming] exten => _XXXXXXX,1,macro(dialuser,SIP/${EXTEN},${EXTEN}) [macro-dialuser] exten =>

Hi all, I'm looking for a reporting solution for Asterisk CDRs. I have a small Asterisk server that will eventually have 4 - 6 trunks. the system is up and the CDRs are being written to a MySQL DB. I tried installing Areski, but had no success .. I assume it's no longer supported... the last update was in March 2005 according to this page..

Hi All, I've just upgraded our CRM and it has an Asterisk Integration Module that I would like to test out. The CRM is running on one of our hosted servers in the cloud. The Asterisk server is running in my office. I am running Asterisk 1.4.21.2~dfsg-1ubuntu3. Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned

I want to install Dahdi and Dahdi-tools on a CentOS 5.3 Xen host and I receive the following error : "You do not appear to have the sources for the 2.6.20-prep kernel installed." I have installed : - kernel-headers-2.6.18-128.4.1.el5.x86_64 - kernel-devel-2.6.18-128.4.1.el5.x86_64 - kernel-xen-devel-2.6.18-128.4.1.el5.x86_64 bash-3.2# uname -r 2.6.20-prep bash-3.2# ls -l

Folks, I've scoured the website and googled, but can't find a definitive answer: What's the difference? OK, the site says 1.4.26.1 is latest stable. Site also says 1.6.2.0-beta4 is latest beta. So what are the others? TIA, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto Visit my blog at:

Hello Please advice how can i configure a sip phone that is not on my local network. ie i have Xlite far some where in America and my Asterisk server is at Sahara desert . Now how can i make a call to that sip phone? Please advice what keywords to carry on?? -- Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello,

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

I am hoping maybe some of you have come across these before in your experience with web meetme. Below are the messages im receiveing when I load the web meetme home page. Notice: Undefined variable: s in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 9 Notice: Undefined variable: logoff_section in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 12 Notice:

Hi, It seems Asterisk needs to be notified that log rotation happened tough applications like astmanproxy or FOP doesn't need to be restarted (nor notified of any rotation). Is this personal observation true ? How could this be explained ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: