similar to: Dyn Exten

Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom --------------

Hi, I successfully installed postgres and prepaid application in my asterisk box but after I digited the code I receive this error: ERROR: Function asterisk_authenticate("unknown", "unknown") does not exist Unable to identify a function that satisfies the given argument types You may need to add explicit typecasts -- Playing 'prepaid-no-aaa'

I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki mailboxnumber@context ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a

Hi, As you asked, I have included my diff to what I did for the DIAL command. I probably didn't stick to some * pre-agreed standard of coding or something, so if these things offend you then I suggest that you close your eyes. :) The biggest thing to consider when you are doing a prepaid system is, what if the person with the same account in/out calls twice? I chose, for now, just to keep

I am trying to install the Modified Prepaid App. I have installed PostgeSQL, created the tables, etc. Make Install runs ok. The when I try to launch asterisk (asterisk -vvvvvgc), it fails to run. I get the following errors, 1st error: [app_prepaid.so] => (Prepaid Application) == Parsing '/etc/asterisk/prepaid.conf': Found Jun 16 14:27:27 ERROR[-1085267840]: app_prepaid.c:127

if i have this setup: [sip users] -- [asterisk] --- [as5300] --- [pstn] asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the asterisk so sip users can call out to pstn. what i would like to is do prepaid on those trunks, not on the sip users. sip users can call any other sip users . i want to do it that way coz i'm trying to build a multi-tenant pbx, and i will use

hai there, today i tried to implement the prepaid application to my * box. I do the step that mantion in to voip-info. i copy the app_prepaid.c and Make file to my asteris/apps, then i run the make. but it show an error like : gcc -shared -Xlinker -x -o app_prepaid.so app_prepaid.o -lpq /usr/bin/ld: cannot find -lpq collect2: ld returned 1 exit status make[1]: *** [app_prepaid.so] Error 1

I'm looking for a prepaid calling card platform that: * easily scales to multiple servers with a common database for: redundancy, capacity, and performance Looking to start with capacity to handle 100 simultaneous calls and be able to easily scale to 1000+ simultaneous calls. * in addition to the normal anti-fraud measures, supports an API for easily adding new anti-fraud tests along the

Hi Guys. I've had a lay off from Asterisk for 12 months but I am starting to look into it again. I am not very Linux savvy and found it hard going the last time. I've started playing with it in the last 3 weeks and I have to admit to making more head way this time. The first problem I'm stuck on and I cant find a solution to is that sound files that I have recorded (be it by

how can we integrate the modified-prepaid application with asterisk because when i compile the app_prepaid it gives bunch of errors.

How do you prepend. I want to be able to dial 7 digits instead of of 11 for local calls. Can someone post there extensions.conf part that is relavent?

Hi, Has anyone done any AGI scripting in pascal. I would appreciate help anyone can offer. My understandin on AGI scripting is very flaky, I am assuming whatever language is used the application needs to be compile and made executable. So if I write a script in pascal, I would compile it with something like freepascal and make it executable. Thanks Umar Sear

Hi I've need to reinstall my asterisk software (hard drive failure). I'm back and running to a make samples state. I have backed up all of my conf files (ok so they were about a week old but much better than starting from scratch), the problem I am having is with WS_FTP Pro. Basically I used to connect to my asterisk server using this software no problems just using root as

Hi All, I hope someone can help. I have a system that I have recently upgraded to latest CVS and my voicemail is not working from mysql database. I get an error on the console saying " No entry in voicemail config file for 'number'" whilst there is an entry in the database for the specified number. It seems like app_voicemail is no longer checking the database even though

Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P -- regards Vikram (http://www.vicramresearch.com)

Hi, i have a problem with iax2 and ringtone. Here is the call path pstn -> asterisk -> iax -> firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really

I am having trouble building appradius from http://appradius.minitelecom.org/ I configure, make, make install cpprad-1.0, but when I configure, then make appradius I get :- obelix:/usr/src/appradius/appradius1.0 # make make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib' make[1]: Nothing to be done for `all'. make[1]: Leaving directory

Hi I have looked through the wiki and search the mailing list, but I cannot find a way to intrude on a call, can asterisk do this feature? if so how? Thanks for your help Robb

Hi, I am trying to implement a dialplan in which the user is notified of a missed call, if no voicemail is left. Basically what I would like to achieve is something like ... exten => _0207XXXXXXX,1,DIAL(SIP/${EXTEN},15) exten => _0207XXXXXXX,2,HasNewVoicemail(${EXTEN:4}@default:INBOX|msgcount) exten => _0207XXXXXXX,3,Voicemail(u${EXTEN:4}) exten =>

Hi List, Forgive me if this has already been covered. I did go through past messages but could not find anything. I want to setup a queue like scenario where users don't need to login/logout. Basically I want to define a list of extensions that will be rung when a call comes in. The sequence in which the extension are rung needs to be intelligent, in the way queues are. For example, it