similar to: login-logout asterisk

Hi, We are trying to implement a complex business logic in Asterisk. Executing "Wait_For_Digit" command after playing IVR. We want to stop the IVR once we receive the digit. It is not recognizing the Digit until it completes the IVR. How can we stop the IVR once we receive the digit? Thanks BB -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Previously i was using asterisk 1.4 with freepbx installation. To try the 1.6 version i installd anc configured everything.. Just one thing didnt work so far.. I am using grandstream 2000 and it has a line busy indicator for chef secretary phones. But now, this feature does not work. I can see the line is online..with a green steady light.. But when the line is busy or DND, it wont change to

Hi Gang, I'm testing 1.8.0 on one of my machines and this snippet "chokes" on line 7 (works fine with 1.4.30) [tb-account-balance] exten => s,1,Set(BALCOUNT=0) exten => s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} )) exten => s,n(runagi),Set(TEST_RETURN="NONE") exten =>

Hello, I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in registering users. As i see from debug it successfully reads from users.conf but later,when a user tries to logon it say peer not found.... And there were an error msg about mysql about the username field..Smthing changed in mysql tables??? Now i downgraded to 1.6.0.9 again and everything is working..

Hey guys, I have stdexten macro dialplan and I have to handle those who doesn't have voicemail box setup. Right now if someone call and if person unavailable the it's just hangup that call. I want it say "person doest have vm setup yet." smthing like that. How should I handle this in my dialplan ? -- Sent from my iPhone

First I wanted to thank both Marc Schwartz Greg Snow and for their reply. Then I needed to add a level of complexity to the problem. I would be able to create the biggest possible matrix. In other way does it exist a method to ask smthing like the following : max number of rows for a matrix if column=x? Thank you ------------------------------------------------------ Passa a Infostrada.

Hello, I'd like to classify data with kmeans algorithm. In my case, I should get 2 clusters in output. Here is my data colCandInd colCandMed 1 82 2950.5 2 83 1831.5 3 1192 2899.0 4 1193 2103.5 The first cluster is the two first lines the 2nd cluster is the two last lines Here is the code: x = colCandList$colCandInd y = colCandList$colCandMed m = matrix(c(x, y),

Hello, I have got a data frame df like this : > df e1 e2 e3 e4 1 1 11 1 21 2 2 12 2 22 3 3 13 3 23 4 4 14 4 24 5 5 15 5 25 6 6 16 6 26 7 7 17 7 27 8 8 18 8 28 9 9 19 9 29 10 10 20 10 30 where e1 ... e3 are vectors I have to select columns which median is in the interval [5;6] ( Here, for instance, e1 and e3) I would not want to use loop... (for, while) I search

Hello, I am trying to enable call forwarding feature on asterisk 1.6.0.9 with asterisk-gui. Sure there is no menu for that on gui but, when i try to write some example scripts to extensions.conf to make it work. I totally failed. I dont wanna install smthing like freepbx etc on the system so, i need help to add unconditional etc call forwarding feature for 1.6 Thanks

Hello, Well.... I need help to do smthing with Samba 2.0.3-19990228 Let me explain... Ive two servers : one NT and one Linux... Each servers control one domain : ==> NT server MYNT : control DOM1 ==> Lin. server MYLIN : control DOM2 when I configure smb.conf like other classics smb.conf files, I can see my server on network explorer on my NT workstation named NTWKS (global network

All, I have been beating my head against a wall trying to figure out how I would implement a separate moderator code and participant code for the same conference using meetme, the deal is I dont want the participants to be able to join until the moderator is in the conference. Is it possible to do this using the apps as they are , or is their a way to use an Agi script, is that the only way?

I need to place a call using a "pin code". To access an external line, the host PBX (a Ericsson MD-110) will require that I dial *72*pincode#phone_number to complete any (trunk) call. When I send the number, my Sipura 3000 will reject the call with "Forbidden - wrong password on authentication for INVITE" (see below). All other calls sent to the Sipura box without the

Hi all, This is to let you know that I found out how to automate the agentcallback logon and logoff. Only thing you need, is to have the agentcode and pincode available in channelvariables. I've updated the documentation on voip-info to incorporate my findings. http://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin Have fun with it! Regards, Michel Koenen

Hi, We currently testing a trixbox/asterisk installation and have used Freepbx to set-up and configure the box and it is running tremendously well. We have an generic IVR configured to which can transfer callers to a child IVR. This child IVR has a number of options to send the caller off to various queues. However we would like to protect some of the options with a password/pin number so that

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

Hello, I am trying to create a simple IVR for testing.. What i did is to create a voice file from asterisk-gui. And i saw it created that under /var/lib/asterisk/sounds/record/ as deneme.gsm Then i tried to make a IVR menu and play that file. I tried exten=s,4,Playback(/record/deneme.gsm) exten=s,4,Playback(record/deneme.gsm) exten=s,4,Playback(deneme.gsm)

Hi, I am using asterisk 1.6.0.10 For debugging i set verbosity to 10 with asterisk -vvvvvvvvvvr.. now i am trying to set it lower but.. when i type asterisk -r it starts with Connected to Asterisk 1.6.0.10 currently running on asterisk1 (pid = 2408) Verbosity is at least 10 when i try set verobisty 1 or similar commands.. i think this command is obselete in 1.6 .. set verbose 1 No such command

I'm just trying to make a real simple Survey via php. Just want it to play the Question Files, wait for a response, save the response into the correct variable and then email it all. I have no issue playing the audio or emailing. But I can't get it to wait for digits or to properly capture those digits into the variables. I know the code is technically right since the emails have this

Hello, for a long time i am using asterisk 1.6 with astgui. but for production system i intend to use asterisk 1.4 which i think might be more robust. And for a more developed service options i preferd to install with freepbx. But still there are big plusses and minusses for both system. My complain about astgui+1.6 was.. For example there were no backup trunk config running on that version.Even

Hello, I want to make an agi script to match incoming DIDs with usernames. I tried to do such entry in incoming trunk. [DID_diddw] include = from-didww [from-didww] exten = 3130XXXXXXX,1,AGI("did.php") exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20) but when i run the rule it says chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to extension