similar to: Playback/background audio from MySQL BLOB

On Mon, Jun 6, 2011 at 10:10 AM, Steve Edwards <asterisk.org at sedwards.com>wrote: > On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards <asterisk.org at sedwards.com> >> wrote: >> >> I strongly suggest using an existing library for the language of your >> choice. >> > > On Mon, 6 Jun 2011, A E [Gmail] wrote: > > Copy that. Not planning to

Has anyone found a good wayt o do a gate intercom using Asterisk? I am looking at a Xorcom PBX with programmable contact, so I have no issue with opening the gate, but the interface at the gate is a bit tricky. I thought about a weather proof housing containing a phone but it seems a bit tacky. I also looked at a handsfree erather proof phone, but at $600 it is a bit steep. Any solutions that

We i have settup it like this it giveme caller id agent id and date-time on gsm file but i want them to be in folder on every day basis datewise. exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb) Any Idea ? Faisal > ------------------------------ > > Message: 16 >

On Wed, 10 May 2017, J Montoya or A J Stiles wrote: > Presumably your staff carry mobile phones. What about an app that gets > the ID of the cell tower to which it is connected, and passes it and the > SIM number in a HTTP request to a server you control? The problem is that they are supposed to use the 'site landline' to confirm presence -- not their cell phone with the

Just and FYI. In RPM form, works on 4.x and I will try on 5.x too. Speech is tacky, but if I can get something but the Gnome Festival synthesizer, maybe it will improve? http://www.adobe.com/products/acrobat/readstep2.html?promoid=BONRM -- Bill

using file_column plugin, if I upload IMG0001.JPG, I get files similarly named IMG0001-thumb.JPG, etc. but if I upload img00002.jpg, I get files img00002-thumb.jpg, etc. This is tacky in terms of my counting on a consistency when implementing views. My inclination is to start hacking away at this section of file_column.rb: class PermanentUploadedFile < RealUploadedFile # :nodoc: def

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

Hi all, I am trying to make calls to R from an MFC application running on XP and am having problems blocking the application while the call executes. I have tried the following approaches to using R from the application (note that I set a wait cursor while R is executing). 1) call rcmd in BATCH mode using system(). This works well, except that I get the cmd window popping up... which makes the

One of my clients is complaining that their customers that use U-verse (and other cable providers) for telephone service cannot enter credit card numbers reliably. The issue not all digits are received in my dialplan. The calls come in on PRI. It's an old 1.2 install, so the only tweak available is 'relaxdtmf.' Any clues on how to proceed? Would jumping to 1.6 help? -- Thanks

I have a 'time and attendance' application. Think janitorial or security kind of thing where an employee goes from location to location. They're supposed to 'clock in' when they get to a site using a phone at that site to prove they're there. Some employees have discovered 'fake caller ID' services can be used to say they're on site when they are not. How

On Tue, Feb 18, 2003 at 09:06:16PM -0500, Jean-Marc Valin wrote: > BTW, when you have something working and stable, I could include it in > the main Speex distribution. Hmmm, define working and stable :) <braindump topic="speexcat"> It began as a merge between speexdec and speexenc from 1.0beta3, with the encoding/decoding removed, and simply piped in and out from ogg

I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality. I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing "alaw", (G.711 A-law) which is the native codec used within the PSTN in this country,

Thanks Steve your work around works great. To ASTERISK.org moderator - If possible I would like to submit this as a feature request. Thanks! On Sat, 27 Feb 2010, Matthew A Kolberg wrote: > I was surprised to find that you can not override the default voice > prompts when using the Authenticate Command. I have viewed the source > and the prompt file names are hard coded. I am

On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'

is typing his number, though there is a 15 seconds timeout, and even if = I type the number very fast it still may happen to me.<o:p></o:p></p><p = class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>It has = been my casual observation that the speed at which I enter digits on my = phone is unrelated to the speed at which my

I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"

Dne 27.1.2016 v 13:14 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> hi, >> >> i have strange problem with asterisk 13 mixmonitor, recording to wav >> (centos6) >> when the system is under load, there are sometimes missing syllable >> >> there arent BIG spikes on cpus >> recordings are to ramdisk (/dev/shm) >>

with the slip of a finger, i destroyed by extensions.conf (grep -i > extensions.conf) I have a backup that is dozens of hours of code old. is there a way i can use the asterisk cli (or some other asterisky method) to recreate that extensions.conf ?

I see the following on one of my new servers: -ts10::sedwards:~$ cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 2979045 2988620 87780075 87779501 IO-APIC-edge timer 1: 1 3 2 3 IO-APIC-edge i8042 8: 0 0 0 1 IO-APIC-edge rtc 9: 0 0 0

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call