similar to: 30 Great free Asterisk applications

Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP

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

I've written another round up article on soft phones that you can use with Asterisk. I've tried to omit any that required you to sign up before downloading, or had time limits to usage etc. There are a couple in there that actually have their own servers but may be interesting nonetheless. I also noticed while writing it up there's a version of Firefly for Mac (beta) now.

Hi, We've just completed an interview with John Todd - the Open Source Community Director for Asterisk. Some interesting comments in it - hope you enjoy it: http://www.venturevoip.com/news.php?rssid=2200 -- Cheers, Matt Riddell Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php

Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface.

Hi, Last night I was working on trying to get Fax for Asterisk installed. I followed all the instructions, but noted that the download selector only provider an X86 32 Bit binary. The machine I'm using is: 2.6.26-1-amd64 #1 SMP Fri Mar 13 17:46:45 UTC 2009 x86_64 GNU/Linux Consequently, on load it presents an error: [Sep 12 10:25:35] WARNING[31306]: loader.c:368 load_dynamic_module:

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

Hi, We're using the Pickup app on a customer's site with Linksys phones. Is there any way to display the callerid of the phone call you've picked up in 1.4? I assume (rightly or wrongly) that this is connected line ID. Basically, the phones are displaying 79 on the screen (the number the dial for pickup) - as you'd expect, but they'd like to see the CID of the person who

Hello, I followed the steps by Russell* http://www.venturevoip.com/news.php?rssid=1980* and I got it working for publish_event only. As soon as I add subscribe_event, Asterisk doesn't start and I just get the following message: *Oct 11 6:38:04.340485 [CLM ] nodeget: trying to find node ffffffff* I have no idea what's wrong. There is not very much information about this issue. Where

I was wondering whether there are any other collections of Asterisk- compatible prompts recorded by Allison, that people might have kindly made available for free download. I found mention of some in an article at http://www.venturevoip.com/news.php?rssid=2690 but the links referred to on www.asterisk.org appear no longer to exist. I already have the core- (good quality) and extra- (poor

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

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 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

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, 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