similar to: Asterisk RPM repo?

Since it is Saturday evening (7PM EST) I am asking this on the list in case someone who knows sees it and can answer. Astricon is in my back yard for the first time, and I could hit you with a rock. I would always like to attend, and spoke at the 2007 Astricon in Phoenix but don't have the idle cycles. Question: Can I just go to Astricon and take the dCAP exam only? In and out? Cost? I

Just a little questions, what's the difference between asterisk 1.8 and asterisk 11? Best regards.

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",

Hi All, I have some problem with Asterisk 1.8 and DIal() to SIP unreachable friend. My dialplan: exten => _XXXX,1,Dial(SIP/${EXTEN},60,rt) Now, when I Dial extension 1050, and there is no 1050 peer registered I got: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument In 1.6 there was no problem, I have got Channel is

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show <queue_name> I get the following numbers: <queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s So from that data we look at 17s holdtime And assume that is the

Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack -- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s at

Hi all. I've got a perl script that connects to Asterisk's management interface using Asterisk::AMI. So far, its proven to be very useful. I'm hoping to use this to detect and respond to asterisk restarts and sip reloads. However, my script gets disconnected quite frequently, causing false alarms in my monitoring. Here's what the code looks like:

Hello, A couple of questions in relation with Asterisk 12 on Debian Wheezy. 1. Can paquet libpjproject-dev (from wheezy-backport) be installed as the sole binary to add PJSIP stack to Asterisk 12 (compiled from source) ? 2. When compiling PJPROJECT from source (see https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject?src=search)), where should PJSIP .so files be located

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

Hey Guys! I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has "reload" command but other doesn't ? satish-desktop*CLI> core show version Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC satish-desktop*CLI> re <tab><tab> realtime reload shirley*CLI> core show version Asterisk

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression

Hi All, I'm having an issue where Asterisk continuously sends out a GAZILLION "SIP NOTIFY" messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine. I'd like to confirm the layout of the

Hi Everyone, I want to know each and every parameter's detail that can be included in the read= write= in manager.conf Where can I find this? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110107/eacb9f1a/attachment.htm>

Hello, are there any deb packages for Ubuntu 12.04? The repos at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packagesare for older Ubuntu versions, also Asterisk 10 is only mentioned for YUM / CentOS? Thanks :-) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Team, Thanks for your great job an Asterisk new features developments. I installed asterisk-12 Beta and found some changes as well which i notice to put in-front of your knowledge, don't know that bug of new modification into objects or old version (asterisk-11) mistake corrected that time, anyway *Asterisk-12:* Array ( [Event] => ConfbridgeMute [Privilege] => call,all [Conference]

Hello Everyone, Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup. After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected. Your help is greatly

Short question: how to get control or notification (gosub, macro, AGI) when a queue member's phone starts ringing due to an incoming call from the queue. Backround: Our phone operators serve both an asterisk call queue and a queue for web chat support. I have a gosub on the queue that calls to our app server to mark the operator unavailable for web chat as soon as they answer an

Hello Guys, In the case of a multiserver environment for outbound automatic calls, can you share you experience and preference between call files and ami originate ? thanks -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -------------- next part -------------- An HTML attachment was scrubbed... URL: