similar to: Forking AGI or GoSub

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking

Hi, Does anyone know of any projects that would allow you to use Redis in place of AstDB? By in place of I don't mean for what Asterisk needs but to store values. For instance for CNAM currently we need to use an AGI to connect to redis to pull CNAM. So in place of: Set(CALLERID(name)=${DB(CNAM/${CALLERID(num)})} it would be done with redis for example:

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

Hi, I have a dial plan where I need to notify an external system when a call was answered and when the call hung up. In both requests the start time needs to be the same. My Dialplan looks something like this: [outbound] Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier)) Exten => h,1,NoOp(ANSWERED_TIME: ${ANSWEREDTIME} >>> DIAL_TIME: ${DIALEDTIME}

Jon, We are only using FastAgi. On the second system (running Asterisk 16) there are no agi's running (just some bash scripts on call hangup). I did add some hackey code (netstat -nua | grep -v 'udp 0 0' | grep -v udp6 | grep -v ' 0 0.0.0.0' | grep udp) to my bash script to check out the packet queue (with the help of

Hi, I am trying to write a dialplan that will use Dial() to call two local extensions. One extension will run an AGI script (a continuous background process, running until hangup), the other will connect the active channel to Jack() (also running as continuous process, until hangup). This is my current dialplan attempt: --------------------- [from-internal] exten = 514316XXXX,1,Answer() same

Hi Dovid Yes was one of the options but as the required list is dynamic becomes very messy - and all combinations problem - where as "call all workers job xxx" is what is needed so the ability to call 20+ numbers is what is needed - agi does a database search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum

Hello all, I've got dialout= and callback= set in my voicemail.conf so that I can have users return calls to folks who have left messages. They really like this feature. But when the callback is over, a normal hangup occurs instead of the caller being put back into voicemail at the next message. Is it possible that the users be returned into the voicemail system where they left off? thanks

Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

Glenn, I can't use MySQL as each node currently has MySQL however there is a lot of data that is stored locally on each box. I may have to take this route if I can't find something else but that would mean syncing all sorts of data that does not need to be synced. On Tue, Jul 30, 2019 at 10:03 PM Glenn Geller (VDOPh) <ggeller at vdo-ph.com> wrote: > Hi Dovid, > >

Hi, I am facing a problem where for legal obligations (LI) I have to copy/mirror/forward the RTP streams for some selected call to an external address/port and I have not found a way to do it with built-in functionality. Do I miss something? The basic requirements are: * Raw RTP (no transcoding, header and payload as is) * Direction (did it arrive at asterisk or was it sent) * End

I know that hangup handlers ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers) have to finish quickly. So it's no surprise that my speech to text agi which takes 8 seconds gets killed. However, can anyone think of a way round this? So, once the caller has hung up, I need to take one of the channel variables, and pass it to a python agi script which then does speech to text.

On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > Doug, > > I tried that as well. Even with my dialplan looking like this: > > > > Ordering by includes works for me under Asterisk 11 and 13 > > What does the output of the below show? > > dialplan show from-external > >

Dovid Bender writes: > The tcpdump that you are running is on the Asterisk box or via port > mirroring? It's on the asterisk box itself. I've already replaced the network card - no change. Thanks, Roel > Regards, > > Dovid > > -----Original Message----- > From: Roel van Meer <roel at 1afa.com> > Sender: asterisk-users-bounces at

Hi, I am running several Asterisk boxes with realtime around the world. Does anyone have a recommendation for a "light" db that would work with Asterisk that would also allow replication between all sites (so if I add an entry to one box it will work with the rest)? TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

Hi, I have an Asterisk box which has an IVR that plays random gsm files. The box has SSD's and two CPU E5-2695 v2 cpus with 64GB ram. The Asterisk CPU usage along with the load seems to jump around. With about 500 callers it hovers between 250-400% CPU (so 2.5 to 4 cores) which seems reasonable. Every so often the load average spikes. The idle never drops below 85%. When the load average

John, That is correct. I am trying to figure out why Asterisk is executing the set part of the execif, if it's coming back as false. On Thu, Feb 13, 2020 at 2:10 PM John Kiniston <johnkiniston at gmail.com> wrote: > My Apologies Dovid, I think I misunderstood your request. > > You don't have the time you need to convert in the format of date string, > Instead you

On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote: > >> Hi, >> >> We have a box up and we are starting to see a lot of "Exceptionally long >> queue length queuing" in the logs. From all the research so far it seems >> like this leads to