similar to: Anyway to know when a channel is going to hangup if Dial Timeout option is used?

Hello everybody. I have a provider that has 3 asterisk boxes which I must balance my calls against. At the moment, I route different destinations to different boxes but this causes lots of problems. Without resorting to OpenSER or other proxies (as my provider also uses IAX), is there a way I can load balance outgoing channels in Asterisk? For example an IAX peer like: [iax_provider] type=peer

Now that the g729 patents have expired, how do we use g729 in Asterisk? Will Digium be releasing a g729 codec for 'free' use or do we download the 'free' codec off the Internet now that we can use it without moral or legal restrictions? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com

> On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards <asterisk.org at sedwards.com> wrote: > Now that the g729 patents have expired, how do we use g729 in > Asterisk? > > Will Digium be releasing a g729 codec for 'free' use or do we > download the 'free' codec off the Internet now that we can use it > without moral or legal

Hi Guys, I am looking to use System() function along with some bash scripting to determine if a Trunk is being used during certain time of the day or not. Here is what I have in mind. Please guide me if you know a better way: exten => s,1,answer exten => s,n,System(/tmp/check.sh) check.sh: check EPOCH time => do an IF for certain times => Allow mutiple calls in certain times and

Hi, I am running asterisk 1.6.1.6 with a howler screamer card. I have g729 and alaw trunks from a pbx /sip providers. The howler screamer will only transcode from g729 to alaw / ulaw but my voice prompts are in SLIN and throws errors when i try and access these applications. Is it simply a case of converting the prompts into other codecs and asterisk will pick these up? ? Thanks

I used to use the local channel to create a global variable (dialplan) [default] exten => s,1,Set(GLOBAL(LSESSION)=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)}) to that end, I modified cli.conf [startup_commands] originate local/s extension s at default = yes But now I upgraded to Asterisk18 and there is no longer a local channels Does anybody have any idea of a workaround?

Hi Everyone, I want to know if a specific codec type is used at least one. For example, I want to know if out of the 100 calls on the system if there is a 1 channel that is running G.729 codec right now. If using dial-plan and I dial in, I can use this to obtain information about CURRENT channel. But it won't allow me to obtain information about OTHER channels and that is what I want to do. I

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}

I have two problems. The first is that when I dial my number from a phone on the Internet or any phone outside my LAN, Asterisk does not respond in any way, which means somehow my system is not picking up the fact that there's an incoming call to it. The second problem is that I thought I'd try an internal phone to see if I could get the hello-world stuff working at the least. I

Hi. I wanted to write a "popcorn" app for myself, both to learn how to script in extensions.conf, but also because it was something handy. Along the way, I found myself doing something like: [popcorn] exten => s,1,Set(FUTURETIME=$[${EPOCH} + 10]) ... exten => s,n,While(${EPOCH} < ${FUTURETIME}) exten => s,n,Wait(0.01) exten => s,n,EndWhile() exten => s,n,Play(beep)

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

On 1.6.1, I must be losing my eyesight: [internal] include => outbound-pstn ............. include => meetme ; 2663 include => setup-meetme-conf-room ; 6000xxxYYYY [setup-meetme-conf-room] exten => _6000XXXNXXX,n,Set(Time-in-secs="${STRFTIME(${EPOCH},,%s}" ) ........ CLI: -- Starting simple switch on 'DAHDI/1-1' [2009-05-17 14:54:49]

So... I'm trying to access CDR(duration) and CDR(billsec) inside h... I keep getting 0. Can I access the CDR function inside a hangup extensions? Asterisk 1.4.13 Thanks, Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

Hi Steve, Thanks for the answer. Since that's what we already have configured, any idea why it wouldn't work? As I said, when "asterisk -rx 'logger reload'" is run it still rotates the log file. On Wed, 20 May 2020 at 18:37, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Wed, 20 May 2020, David Cunningham wrote: > > > We have an Asterisk

Dears I already read most of post on asterisk group and (http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning) But I could not find a calculator 1-Is there a calculator I can download for that 2-What I the maximum simultaneous calls that can asterisk handle using CPU 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding , 3-And what is the number of

Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels. I want to use the teleyapper system for broadcasting call for security reason but i need that all channels are free when a security call is ready to start! I already search in the old post without success. Can anyone help me? Thanks and sorry for my newbie english -------------- next part -------------- An HTML

Hi, I've been trying to get MoH files to sound decent. I've got a hold of Royalty-free Classical music (a safe choice for most of my customers) and I`ve been trying to convert them to the normal telephony/Asterisk format using sox. Unfortunately, it sounds really bad. I don't expect concert hall quality of course, 8000KHz being what it is, but is there a better way to convert

Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the "too many file" error, and fixing it using ulimit..... Also, is there any way we can allocate sufficient

Hello All, one of the extensions fall into a loop, I don't know how to hangup that channel -- Executing [i at autoatten:2] Goto("Local/100 at sipphones-000001b2;2", "s,2") in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200 at sipphones-000001b2;2 -- Executing [i at autoatten:1]

Hi again! I always try to get my mobile phone work with my Asterisk. I tried to install Asterisk on my PC (with public IP), but it has problems, too... I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider does not want it, too, since I have no problem to connect and get a very good audio quality if I connect to other SIP-Provider or to an Asterisk (SAME