similar to: Getting calee audio in Asterisk (real time)

Hi, I have been headbanging with asterisk and Jack for a while, decited to ask other linuxists for an advice. The problem is that Jack is compiled from source (0.118) in /usr/local/, but menuselect says "XXX" for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide

Hello asterisk users! I, like many people, have a cell phone. I also have some SIP phone devices (software and hardware). I'd like to have one number that rings all my phones and routes the call to wherever I pick up. However, my cell phone has its own call forwarding voicemail. I can't just turn that off, because then direct-to-cell calls wouldn't ever get to voicemail - that

Hi, Thanks for your info, What is the impact of the following line in dialplan, Dial(SIP/19201/19202,300) On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes <vinicius at aittelecom.com.br> wrote: > You might want to use the Originate() application instead. Check its usage > by issuing the command 'core show application originate' on Asterisk CLI. > > 2015-09-03

Hi, List, I am looking for a cheapest (and therefore most funny) way to attach GSM card to my asterisk home box. Needed features: Calls+SMS in/out one or two SIM cards (ports) Should I try looking for a GSM PCI card that is compatible with linux/asterisk, or GSM USB card, or modern full-blown SIP GSM gateway (with ethernet)? Maybe an ordinary cell phone with USB interface and mangling with

Hi List, Our company has several small distributed offices we would like to inter-connect with bridged VPN a single subnet (last example in http://www.shorewall.net/OPENVPN.html). We have SIP phones in every office (up to 5) so we can use SIP without any NATing and securely. Max theoretical simultaneous calls possible ~30, but we have ~5-10 @ regular basis. OpenVPN server would be in the same

Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and "This user is temporarily unavailable". Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio

Hi, my doubt is very simple. I'm sure I've seen someone using something like this before, but unfortunatelly my searches in the archives were useless. Well, I have some objects called after a name that has a number attached to it, varying. Let's say I have: > ls poly1 poly2 poly3 poly4 poly5 poly6 ... I would like to access these objects using a for(), in which I could do

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hello there, I have successfully installed and configured asterisk for use as an office PBX using SIP trucks and Voip handsets (using g.729 codec) which works great. Now I wish to try and configure asterisk to do a HTTP request and submit callerID to an external website when a call is missed. eg Someone calls PBX and rings extension 100 -> Call is not answered -> HTTP request is initiated

Dear List, My Dial command: exten => _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1)) exten => h,1,.... [connect-jack] exten => _X.,1,NoOp(${CHANNEL}) ; Leg A exten => _X.,2,NoOp(${CHANNEL}) ; Leg B The problem is: after answering, [connect-jack] both priorities are executed, and right after executing them call drops. Log: -- Executing [123456 at NPDB2:76]

Hello, Asterisk party, If block the call before dialing (Hangup()), CDR's don't write to MySQL or CSV. Otherwise, if Hangup() is after Dial(), CDRs write normally. Here is the dialplan: ; we skipped dial, because the number is "blocked" exten => _X.,n(Finish),Hangup() exten => h,1,NoOP("hangup") exten => h,2,ResetCDR(w) exten => h,n,NoCDR() exten =>

Hi, Is there a way to check on AMI if a user is currently engage on the phone? i would like to display on my portal whether a user is calling or not. thank you regards Ron

Hello list, I'm using asterisk 1.4.30 and realtime sip. I notice that the field "musiconhold" is not working as when putting someone on hold, the default musiconhold class is always used. musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes my realtime sip peers have the

On Aug 7, 2007 'Mojo' wrote: Nicholas Blasgen wrote: > I've got 4 SIP phone lines with a call-limit of 2 for each. I've > written a handy macro to allow my users to dial a phone number and the > macro will figure out the next available line to use by first checking > if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a > backup, and if it

Hi, I have an xlite registered with asterisk server. When i dial a number AGI is invoked. and in this we are running to threads one to record files and one to play files. So i dialed the extension and i started recording and playing at the same time. On the xlite i m getting an indication when recording my voice and at the same time i could see playing the other file too. But in the directory

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

Dear list: I am trying to calculate the intersection area between two irregular polygons (see example data below). set.seed(1234) theta <- seq(0, 2 * pi, length=(100)) poly1<- cbind(c(0 + 1 * cos(theta) + rnorm(100, sd=0.1)), c(0 + 2 * sin(theta))) poly2<- cbind(c(0 + 2 * cos(theta) ), c(-1 + 1.5 * sin(theta)+ rnorm(100, sd=0.1))) plot(x, y, type = "n", ,

Hi We have the following test .call file and test dialplan: I can't set a custom CDR var to a value on one channel leg, and another value on the connected channel leg? Is there a way I can woraround this issue? ## test call file Channel: Local/queue at TiagoGeada CallerID: teste-geada:0:210332450: MaxRetries: 0 RetryTime: 1 WaitTime: 8640 Account: teste-geada Context: TiagoGeada

Hello, Using 1.6.2-rc5, my settings include: [local-phone](!) context=mylocal type=friend nat=no canreinvite=no host=dynamic qualify=yes dtmf=info language=fr call-limit=5 subscribecontext=subs disallow=all allow=alaw t38pt_udptl=no setvar=accountcode=foo [168](local-phone) defaultuser=168 secret=pass168 callerid=John Doe<168>

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