similar to: return code of app in dialplan

The Queue app says that queue will return a value (zero or -1). How do I test for this value?

Hmmm.... Can't place calls... Can't access website... Neither of the 3 nameservers answer anything... Anyone heard/know something to explain all this?

Hi, Looking through the source code, I see two functions in the 'libpri' code for message-waiting: pri_mwi_activate, and pri_mwi_deactivate, but they are not used by anything anywhere. Is there a sample application somewhere which makes use of these functions? I believe that the ISDN MWI is used when a Norstar MICS has an external voicemail provider. I don't know exactly how this

Hi, I dropped an asterisk server with a TE405P between a Norstar Meridian PBX and it's PRI PSTN connection. Everything seemed to work fine using a pass-thru-type dialplan configuration... except now we've realised that outbound calls to celphones get dropped upon connect, but not on every call (almost always the first try, but not the second). All other calls to non-celphones had no

Sorry for the crosspost (this was also posted to asterisk-at-uc-dot-org) but I haven't got a response. I have a cell phone added to a queue as a local extension (member => Local/299). I want the cell phone to be able to reject calls to the queue without the person sitting in the queue being hung up on, etc. The way my dialplan is set up, the person hits 1 to answer the call and any other

Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel

Hello, We've had dozens of Polycom 3.x firmware phones deployed and working great for years. Now I've finally been charged with the long-overdue task of figuring out why newer Polycom devices with 4.x firmware register fine but do not respond to SIP OPTIONS request and therefore always become UNREACHABLE if the sip qualify setting is set to yes. To my dismay, searches for solutions from

The meetmeadmin() dialplan function lets you specify a user to mute, un-mute or kick. But how do you get a list of users in your dialplan? When a user joins a conference, the user number assigned is "the last user number +1." If you have a long running conference with callers joining and leaving all the time, this can grow to be a large number. I want to be able to

hi list, I'm looking for a way to execute commands in my dialplan specifically when a caller has hung up. my curretn dialplan looks like this: exten => s,1,Answer exten => s,n(restart),BackGround(intro) exten => s,n,Read(Enter,4,4) exten => s,n,Voicemail(${Enter},u) exten => s,n,agi(process.php|${Enter}) exten => #,1,Playback(thanks) exten => #,n,Hangup it lets a user

Hello everyone, I'm having some troubles with some dialplan logic I've written which sends missed call notifications via e-mail. It's currently sending these notifications even if the call was answered, marking them all as hung-up. What I've been able to see is that the macro never reaches the "s-ANSWER" bits which mark the call as successful. I've posted my

We're having an issue where we call ReceiveFax in a context that includes a hangup extension and half the time dialplan execution doesn't continue after the fax is received successfully. Am I missing something simple here? Below is a sample call where this happened: The last log line for this channel/call is: [Feb 21 09:10:53] VERBOSE[13730] res_fax_digium.c: -- Channel

Hi, It seems that the "switch" parameter has a priority in the dialplan that is higher than the wildcard extensions.. This I am finding to be a problem.. My setup.. UA1--[AST1]--{IAX}--[AST2]--UA2 | | PSTN1 PSTN2 I use switch on AST1 to connect to AST2... As you can see I have PSTN connections on both and also the IAX connection is not permanent.. I

I have a problem where if an agent's extension is busy and has voicemail the queue app will follow the dialplan and send the caller to an agents voicemail. This is really bad, because it takes the caller out of the queue when it hits that agent. But we also would like to have voicemail for some extensions like the shift managers etc. Is there s solution/workaround/patch? Thanks, -Ryan

Hi all, there is an application called changrab with quite interesting capabilities: http://www.freeswitch.org/asterisk_stuff/app_changrab.c I think it is available in the 1.4 version by default!? This application can connect to channels which are already UP. The only possibility AFAIK to connect channels after they are UP are the well known conferencing applications meetme and conference. If

Hi, i have context in a dialplan, I want to "execute" this context without insert the Answer Application (s? ..without call any ext). Example : [sistema-allarmi-principale] exten => s,1,Set(GRUPPO=${DIAL:-2:1}) exten => s,2,Set(ALLARME=${DIAL:1:1}) exten => s,3,AGI(checkgroup.php|${GRUPPO}) ;rest of... I tried with a Call Data File.. i create a CallDataFile like this :

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Hi Asteriskers, I have the following : exten => 1,1,Playback(sample) exten => 1,2,Read(response,,1) exten => 1,3,GotoIf($["${response}" != "*"]?300:100) exten => 1,100,Playback(hello) exten => 1,101, [[[ do stuff ]]] exten => 1,300,Playback(reject) exten => 1,301,Hangup Which plays a confirmation sample, waits for the user to press * and then continues

Dear community, I use puppet to deploy iptables on every server. This is working fine. In the manifest I defined an iptables service which leads to my init script. The thing is each time I run puppet I got : notice: /Stage[main]/Iptables/Service[netfilter]/ensure: ensure changed ''stopped'' to ''running'' I would like to add a status command to my startup script

So, I've kind of converted my dialplan from: exten => custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?acd_one_queue,custcare-open,1) exten => custcare,2,Goto(custcare-closed,1) exten => custcare-open,1.... exten => custcare-open,99.... exten => custcare-closed,1.... exten => custcare-closed,99.... to: exten =>

Hi, I have tried to use the Record Command in Asterisk, Here is the configuration : exten => record,1,Answer ... exten => record,n,Record(/var/spool/asterisk/record/${CALLFILENAME}:WAV) exten => record,n,Playback(vm-goodbye) exten => record,n,system(/usr/local/bin/send-recording.pl --to ${EMAILADDR} --file /var/spool/asterisk/record/${CALLFILENAME}.WAV) exten => record,n,Hangup