Daniel Eboa
2005-Jan-27 04:23 UTC
[Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk
Hello I got the similar error while trying a call. -- Executing Answer("SIP/8000104-86ef", "") in new stack -- Executing Wait("SIP/8000104-86ef", "2") in new stack -- Executing AGI("SIP/8000104-86ef", "areskicc.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php areskicc.php: 'agi_request' => 'areskicc.php' areskicc.php: 'agi_channel' => 'SIP/8000104-86ef' areskicc.php: 'agi_language' => 'en' areskicc.php: 'agi_type' => 'SIP' areskicc.php: 'agi_uniqueid' => '1106824539.3' areskicc.php: 'agi_callerid' => '"DTA-310" <8000104>' areskicc.php: 'agi_dnid' => '002379511272' areskicc.php: 'agi_rdnis' => 'unknown' areskicc.php: 'agi_context' => 'prepaid' areskicc.php: 'agi_extension' => '002379511272' areskicc.php: 'agi_priority' => '3' areskicc.php: 'agi_enhanced' => '0.0' areskicc.php: 'agi_accountcode' => '' areskicc.php: areskicc.php: >> ANSWER areskicc.php: string(56) ""DTA-310" <8000104> ; SIP/8000104-86ef ; 1106824539.3 ; "n -- AGI Script areskicc.php completed, returning 0 -- Executing Wait("SIP/8000104-86ef", "2") in new stack -- Executing Hangup("SIP/8000104-86ef", "") in new stack == Spawn extension (prepaid, 002379511272, 5) exited non-zero on 'SIP/8000104-86ef' Need some help. Thanks Daniel. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Areski Sent: jeudi 27 janvier 2005 11:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk Hi Alex, Concerning the web interface, in this version we need the register_globals = On I will try to change it in the next release... To find out the error on the agi, can you run the agi script manually. php areskicc.php You will get more details about the error! Regards, Areski On Thu, 2005-01-27 at 03:07, Alexander Romanov wrote:> Hi, > > I've tried it and could not get to work any of them (webapp and agi). > > On webapp I do not get a full menu, just "logout" and "disconnect" > With agi nothing happens when I execute the script. > > -- Executing Answer("SIP/2204-6221", "") in new stack > -- Executing Wait("SIP/2204-6221", "2") in new stack > -- Executing AGI("SIP/2204-6221", "areskicc.php") in new stack > -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php > -- AGI Script areskicc.php completed, returning 0 > -- Executing Wait("SIP/2204-6221", "2") in new stack > -- Executing Hangup("SIP/2204-6221", "") in new stack > == Spawn extension (local, 40, 5) exited non-zero on 'SIP/2204-6221' > > > I have followed instructions to the letter. Am I missing something? > > Alex. > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Areski > Sent: Thursday, 27 January 2005 4:05 AM > To: Asterisk-Users Mailing-list > Subject: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application > forAsterisk > > > Hello everyone, > > > If you want to know why I am so tired today :D > Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just > finish it yesterday night! > > > Briefly, AreskiCC is an AGI script and PHP-Web application which greatly > handle the complete CallingCard System. > > > FEATURES - AGI : > * Authenticate with the use of a Cardnumber > the Cardnumber can also be defined as accountcode into sip.conf, > iax.conf, etc.. > * take care of multiple calls using the same Cardnumber > * Caller gets informed about his credit > Announce the remaining credit > * Caller is requested to enter a destination number > * Announce the maximal call time for the given destination number > It calculates the remaining duration of the actual call (based > on tariffrate tables), informs the caller about this and sets a > timeout > * Interupt the call if the card balance gets zero > Warn the caller about the call interupt 60 & 30 seconds before > the call gets interupted > * It connects the Caller to the destination through the configured > trunk > note : different trunks can be configured and associated by > prefix > * After disconnecting the call AGI updates the credit and stores > the concerning Call-Detail-Records with CallingPartyNumber, > CalledPartyNumber, CallSetupTime, Duration, Charge and the > remaining credit > > > FEATURES - WEB INTERFACE: > * CARD/CUSTOMERS > * List customers > * Refill customer > * CARD/CUSTOMERS > * List customers/cards > * Refill customer/card > * Create customer/card > * Generate customers/cards > * BILLING > * View money situation > * View Payment > * Add new Payment > * RATECARD > * List Tariffplan > * Create new Tariffplan > * Define Tariffplan > * TRUNK > * List Trunk > * Add Trunk > * CALL REPORT - BALANCE > > Last note : It's distributed under GNU GPL Licence. > > > > I hope there will have a big interest for the soft, > I am waiting your feedbacks... > > Regards, > /Areski > > > > > > -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ > > Bela?d Arezqui > www.areski.net > E-mail : areski [alt@alt] gmail (.dot.) com > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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