similar to: Use of Storage Area Network with Asterisk

Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150627/6774c750/attachment.html>

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

Is there a recent survey of that Linux distro and version people are using for the Asterisk installations? I recall seeing a pie chart over a year ago (I think on a wiki but I can't find it again)....also hoping for something more current. I suspect RH5 and RH6 are most popular...but I'm looking for facts -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is the a CLI command that shows all channels in use at one time? (Whether IAX, SIP, SCCP, etc)? As well, when I "SIP SHOW CHANNELS" I see phones registering showing as channels in use. Is there a way to filter this output? Thanks! MD

I need to disable/enable a peer after hours automatically, and am thinking about doing so via the AMI. Is there a command to enable/disable (or perhaps delete/add) a peer via the AMI? I could create code to modify sip.conf and force a reload, but that seems like the wrong approach... -------------- next part -------------- An HTML attachment was scrubbed... URL:

?I meant how many calls are in progress on a particular trunk. (Sorry - I didn't even think of the other interpretation). ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Matt Riddell <lists at venturevoip.com> Sent: Sunday, June 28, 2015 9:26 AM To: Asterisk Users List Subject: Re:

I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD

We have an application that plays a variety of sound files on one leg of a call (generated by a call file). We've been told that the party listening to the audio files intermittantly hears "robotic" sounding audio (on/off during the same call). Anyone have ideas on cause? These calls are on an internal network (lots of network bandwidth), and from a server running 99% idle.

Can someone tell me when the /proc/dahdi files are created for spans? Are they created when asterisk starts (or the asterisk init script) - if not what script creates them? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150204/f14ae5a2/attachment.html>

I have a custom voicemail script which reformats and forwards the attached voicemail wav file to the recipient. I would like to make use of a channel variable in my script; is there a way to pass a channel variable to this voicemail script? -------------- next part -------------- An HTML attachment was scrubbed... URL:

After reading about the 2 major SSL (and TLS?) weaknesses discovered this year, I was wondering how it affects asterisk. Does the SIP authentication use TLS - or something that was recently broken? Is there a risk of exposing passwords? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have a multitenant Asterisk 1.4 installation for multiple small business, and we need to report how many calls a single business has active at one time. Is there a way to VIEW how many calls are up in a single context? (Or some other way to accomplish the same)? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I realize this is getting a bit outside myth...but hopefully someone can offer some ideas... I'm using the latest NVIDIA drivers on Fedora 12, with Nvidia 8600GT. Although the dual DVI outputs work great, the driver just won't detect anything connected to the component video connector. Is anyone out there successfully using the component video out on their Nvidia card with a recent

I've searched through the wiki but I can't find what I need...I'm trying to figure out what the max call duation is. I found references to "show application AbsoluteTimeout" but that isn't in 1.6 (not even prepending "core" to the front). A core help show didn't help... -------------- next part -------------- An HTML attachment was scrubbed... URL:

When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data)" IAX2/FONEMITEL123456 1296197222 at entryhome<mailto:1296197222 at entryhome> Ringing

I'm creating an AMI client and I only want to get newchannel events (as well as responses to any actions I initiate). What would I set the eventmask to to only get the newchannel events? For anyone else looking...is there a table somewhere online that maps events to their eventmask categories? I checked the asterisk wiki and voip-info but can't find this... -------------- next part

I (canadian) store has a deal on for the vera lite controller: http://www.tigerdirect.ca/applications/searchtools/item-Details.asp?EdpNo=8930107&sku=VEP-STARTER1 but this looks different than the vera lite green & white: http://www.amazon.com/Mi-Casa-Verde-VeraLite-Controller/dp/B007005364/ref=cm_cr_pr_product_top? and maybe there are more zwave controllers. Can someone (who is

I have setup an Ast 11.6 host and I want to login via AJAM. I setup manager.conf, http.conf described in the docs. When I login via the AMI it works fine (see below), but when I login via AJAM the same credentials fail (see further down) Can someone tell me how to fix this? ----------- Connection closed by foreign host. root at pbx:/tmp# telnet localhost 5038 Trying 127.0.0.1... Connected to