similar to: Return code from queue app

I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registration looks like so: /SIP/Registry/3115552368 :64.169.xx.yyy:38836:3600:3115552368:sip:3115552368@64.169.xx.yyy:5060 When the second phone tries to register, it gets back a 404 not found. Not a

Hi, I'll probably kick myself when I read the replies to this... How do I test the return code of an app in the dialplan? I need to test if the app, MYSQL() in this case, returned -1 or 0. It's easy to see after-the-fact in the log output, but I need the result in the dialplan, I just can't find which variable stores the actual return-code. Thanks!

I need to send calls to a bank of servers in a round robin fashion. Has anybody implemented this in dialplan language? If not, in some other fashion. </edg>

I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP,

I'm trying to set up a way that our users can send an XMPP message to Asterisk (unsolicited) to request information, such as voicemail status or the like. No matter what I set for the dialplan, I'm only seeing Asterisk execute the s,1 priority in the context defined in xmpp.conf for incoming messages, and then the "call" hangs up without executing further instructions. Anything

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds after 3082 has started speaking to the outside caller - 3082's call goes dead in their

Our puppetmaster runs 0.24.8 on Ubuntu 9.10. Our clients are either Ubuntu 9.10 or Ubuntu 8.04. We''ve just brought up our first Ubuntu 10.04 machine. This machine installs puppetd 0.25.4. The Ubuntu 10.04 machine can''t seem to present it''s certificate request properly. In my masterhttp.log, I see [2010-10-20 13:09:06] 174-143-141-55.static.cloud-ips.com - -

I'm having a problem with polycom ip601. If I Dial() directly eg Dial(SIP/4000) it works perfectly. The polycom rings, and stops ringing as soon as I hang up. But if the phone is called via a queue, the polycom continues to ring long after I've hung up. Other phones in the queue (grandstream, cisco) don't have this problem, and stop ringing properly when I hang up. polycom bug

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

Hi, Is there a way to execute next priority in the dialplan if you have called agi:async? I want to play warning message if adhearsion is down. Currently I wasn't able to make it work. The dialplan execution ends after the first priority. [incomming] exten => _X.,1,AGI(agi:async) exten => _X.,2,Answer exten => _X.,3,Playback(some-message) exten => _X.,4,Hangup Regards, Pavel

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Hi, We have a "simple" AGI script that provides some IVR functionality. It connects to a MySQL database in order to create a call record and capture IVR data. During the IVR process, we need to store the time the call started, so basically we INSERT a new MySQL row with callstart = NOW(), uniqueid = AGI(agi_uniqueid). As the user selects different options, we update the row to reflect

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

Hi All, I?m trying to do ?things? after my Dial application terminates (e.g. play IVR to called party, calling party, etc.). I?m trying to use the local channel for this purpose but so far with no success. I?m using 1.6.1.18 and this is my extensions.conf: [Internal] exten => _22,1,Dial(Local/${EXTEN}@CW/n) ; 22 is test number exten => _22,2,Noop(After Hangup) [CW] exten =>

Hello List, I'm using a dialstring like the one below. I want to have three different things happening depending on exit cause. Dial(SIP/${phonenumber},20,gL(20000[:5000][:5000])) These 3 things could happen: 1, Caller hangs up 2, Callee hangs up 3, The 20 seconds is up and call is terminated from Asterisk. Is there a way to separate these 3? Thanks, Best regards, Tobias --------------

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

Does anyone know of a speech recognition module (like say yes or no, or numbers) I guess due to the complexity of speech recognition it might just be found in commercial applications or am I wrong like always? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050128/119168cb/attachment.htm

SendImage() in 1.4: ---cut--- SendImage(filename): Sends an image on a channel. If the channel supports image transport but the image send fails, the channel will be hung up. Otherwise, the dialplan continues execution. The option string may contain the following character: 'j' -- jump to priority n+101 if the channel doesn't support image transport This application sets the

Can anybody tell me the difference between a TE410P and a TE405P? Is it JUST the 5v vs 3.3v pcis slot spec, or is there some thing else? </edg>

I am experiencing a change in behaviour of my Queues in 1.4.9 vs 1.4.8. I do not pass the 'n' option to any call to Queue() in my dialplan. Yet since I upgraded to 1.4.9, I have occasionally seen this on my console: -- Nobody picked up in 20000 ms -- Exiting on time-out cycle That log message "Exiting on time-out cycle" is exclusive to the logic in app_queue meant to