similar to: Executing system commands through Manager API

On 4/5/16 3:17 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> I am currently having a voice quality problem with one of our Asterisk >> servers. We have checked the network and we have found no problems that >> could cause the voice to sound cracked and with small interruptions. I >> am looking at the timing source for Asterisk and it is currently using >>

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director

Hi, Is it possible to configure Asterisk to fetch for files from the spool directory in different directories? For example, fetch voicemail files in /abc/voicemail and call files in /cde/outgoing ?. And is it possible to configure the filename that Asterisk gives to files, like voicemail files? Thanks, Ricardo

On Mon, 6 Feb 2017, Tech Support wrote: > We were able to develop a feature to send the call to voicemail about > 90% of the time. That way, an end user could (1) not be bothered by > having to answer the call, (2) delete the message without listening to > it, or (3) listen to the message when it was most convenient for them. > That way, they were in control and things were

> On Mon, 6 Feb 2017, Tech Support wrote: > > We were able to develop a feature to send the call to voicemail about 90% of the time. That way, an end user could (1) not be bothered by having to answer the call, (2) > delete the message without listening to it, or (3) listen to the message when it was most convenient for them. That way, they were in control and things were

Hi guys, I hope I get your help for this problem ... at least point me through the right direction .... I configured a queue that when the people are tired of waiting they can press 1 and I direct them to voicemail ... but I'd like to make a callback solution at this point, because during busy hours it is hard to check the voicemail ... I'd like to make my system to call back this

Not really, doing the way below you don't even have to worry about it. They both go out at the same instant and as soon as it hits voicemail it disconnects the other leg. If you wanted you could leave it ringing for twenty minutes and it would still have the same effect. Kind regards, Matt > On Feb 6, 2017, at 12:29 PM, Tech Support <asterisk at voipbusiness.us> wrote: >

Anyone have a cookbook on configuring sendmail to work with Asterisk? Or,a few config examples.

I had been using google tts, but it started requiring a captcha for my browser, and via linux I can't access http://translate.google.com/translate_tts?q=test (redirects to captcha) as so, its not reliable On 27 August 2015 at 17:16, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 8/26/15 1:15 PM, Tech Support wrote: > > All; > > I have a customer who is

I have a Sipura SPA-841. It's developed a nasty habit. At random times, it likes to dial my cell phone voicemail number and play my messages to anybody who happens to be within earshot. Any clues where to look at what's going on? My voice mail number (extension 220 in my dialplan) is the only number being dialed. When this happens, show channels looks like this: IAX2/NuFone-1

Hi, I'm sure this topic has been discussed before but i'm having trouble finding a simple answer. Whats the easiest way of sending an email from Asterisk? I want to set up a warning so that after a Dial cmd, if the DIALSTATUS is CHANUNAVAIL, Asterisk sends an email to the admin to check the voip phone is connected properly. I've got the dial plan set up, I just dont know what

'lo, So yesterday, one of our clients had the misfortune of having the disk that their Asterisk config (*.conf) was stored on take a dirt nap. Of course, Asterisk was still running at the time, and everything continued to work (except for voicemail, which was stored on the same disk) right up until I shut down Asterisk to investigate what was going on. Because the disk was dead, though, I

Hi all I got this solved. Turns out the script WAS executing, but I forgot that apparently you need to follow "cron rules" in any BASH scripts executed via System() from an Asterisk dialplan. E. g. all paths must be fully and absolutely specified, there are no relative path references available. So I changed the the file wireless.sh which was: #!/bin/bash touch wireless-executed to

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"

I'm presently working on a project to build a scheduling system accessible by both web and phone. on the web side one can query what items are available when by using the time or the item as a key then reserve for an available time slot. reservations may also be modified by the user that made them or an admin. Where may I find examples of doing similar things with asterisk? all I've

Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi all, new to the list and this is probably a basic question and couldn't find anything clear googling around but I don't know how to handle calls to sip extensions not defined on sip.conf while using pattern matching. On my example I have sip extensions 10, 11, 12, and 13 on sip.conf. On a basic extension.conf I set up a pattern starting with "1" and a second digit should dial

I have "externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl" so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the

I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? -- Thanks in advance,