search for: app_agi

...s able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this? Below is a patch that solves the problem. Index: asterisk/apps/app_agi.c =================================================================== RCS file: /usr/cvsroot/asterisk/apps/app_agi.c,v retrieving revision 1.22 dif...

...v,' Jun 22 13:53:06 WARNING[1209214400]: app_record.c:221 record_exec: Could not create file /tmp/asterisk/incident-3893006535 Jun 22 13:53:08 WARNING[1209214400]: file.c:464 ast_openstream: File /tmp/asterisk/incident-3893006535 does not exist in any format Jun 22 13:53:08 WARNING[1209214400]: app_agi.c:336 handle_streamfile: Unable to open /tmp/asterisk/incident-3893006535 == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Any ideas on how to make AGI parse this arguement correctly? Thanks, Tom Daly

Hi, I don't know if it's possible to stop a STREAM FILE pushing a key. Anyone know it? Best. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030507/04d18302/attachment.htm

Some switches provide the functionality to try a number till it becomes available. Thus when one dials a number and get a busy, one enters a *XX# code and the switch will call your extension when the called party becomes available. Has somebody already built this in/for Asterisk, otherwise I'll look into it. Michiel Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT

Hellooo... Is it possible to fill a variable of PHP-based-AGI-script from dialed extension ? This is what I need to achieve: If someone dial an extension, say 777, I want the dialed extension (777) be filled into PHP variable. I need the dialed extension become a condition of PHP script. Help please... Thanks romsun _________________________________________________________ This mail sent

...tivity_alerts/wavs/123456_1_1_0.745781945801 "" 0 WARNING[2160657]: File format_wav_gsm.c, Line 194 (check_header): Does not say data WARNING[2160657]: File file.c, Line 379 (ast_filehelper): Unable to open fd on /activity_alerts/wavs/123456_1_1_0.745781945801 WARNING[2160657]: File app_agi.c, Line 322 (handle_streamfile): Unable to open /activity_alerts/wavs/123456_1_1_0.745781945801 RESULT_LINE: 200 result=0 endpos=0 == Spawn extension (activity-alerts, s, 2) exited non-zero on 'IAX2[NuFone@Outgoing1]/2' At the point where it tries to playback the file the warnings are sp...

Hi to all, I have configured in my * a avm fritz card and in my isdn connection I have only 1 extensions. I read from documentation tht is possible to use the exten,fax to redirect an incoming fax. Is it possible also with avm card or only with digium card. I used in this way but with no success: exten => fax,1,Goto(fax,2202,1) 'Configured for spandsp exten =>

...OFF(0) on Zap/2-1 Jul 2 12:27:39 DEBUG[630802]: File chan_zap.c, Line 908 (update_conf): Updated conferencing on 2, with 0 conference users Jul 2 12:27:39 DEBUG[630802]: File chan_zap.c, Line 1930 (zt_setoption): Set option AUDIO MODE, value: OFF(0) on Zap/2-1 Jul 2 12:27:39 DEBUG[630802]: File app_agi.c, Line 1216 (run_agi): Zap/1-1 hungup Jul 2 12:27:39 DEBUG[630802]: File cdr_mysql.c, Line 58 (mysql_log): cdr_mysql: inserting a CDR record. Jul 2 12:27:39 DEBUG[630802]: File cdr_mysql.c, Line 61 (mysql_log): cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,c...

...m.so load => cdr_csv.so load => pbx_config.so load => pbx_spool.so load => pbx_wilcalu.so load => res_adsi.so load => res_crypto.so load => res_indications.so load => res_monitor.so load => res_musiconhold.so load => res_parking.so load => app_adsiprog.so load => app_agi.so noload => app_alarmreceiver.so load => app_authenticate.so load => app_cdr.so load => app_chanisavail.so load => app_controlplayback.so load => app_cut.so load => app_db.so load => app_dial.so load => app_directory.so load => app_disa.so load => app_echo.so load...

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

...Special Informa 0 app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 app_getcpeid.so Get ADSI CPE ID 0 app_adsiprog.so Asterisk ADSI Programming Application 0 app_qcall.so Call from Queue 0 app_agi.so Asterisk Gateway Interface (AGI) 0 app_disa.so DISA (Direct Inward System Access) Appli 0 app_url.so Send URL Applications 0 app_image.so Image Transmission Application 0 app_record.so T...

A note to all those who are avoiding writing up an AGI becasue it looks two complicated.. I have just written up my first and its awesome.. It makes Asterisk open to all sorts of possibilities.. let your imagination run wild.. I put off writing an AGI script because a) I could not find any docs b) it looked like the only way to do it was perl and I know nothing about perl and c) I am not a

...== Spawn extension (macro-receivecall, s, 2) exited non-zero on 'Zap/2-1' in macro 'receivecall' == Spawn extension (home, 102, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' and these in /var/log/asterisk/messages: Sep 19 00:31:08 WARNING[458770]: File app_agi.c, Line 1277 (run_agi): No channel, no fd? Any idea on what might be wrong ? I have tested these way of doing a follow me routine on 3 different asterisk boxes, with different setups and i got the same result on all of them. The AGI's are suppoused to do a more complex task, like searc...

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

I have been attempting to download, compile and configure spandsp to function with * without much luck. I am guessing that some assumptions were made regarding the users knowledge level of Linux. Sadly, I must not live up to those assumptions. My problem begins when after compiling spandsp I look for the app_rxfax.c, app_txfax.c, app_dtmftotext.c and makefile.patch files to place in the

...(Generic System() application) [app_echo.so] => (Simple Echo Application) [app_record.so] => (Trivial Record Application) [app_image.so] => (Image Transmission Application) [app_url.so] => (Send URL Applications) [app_disa.so] => (DISA (Direct Inward System Access) Application) [app_agi.so] => (Asterisk Gateway Interface (AGI)) [app_qcall.so] => (Call from Queue) [app_adsiprog.so] => (Asterisk ADSI Programming Application) [app_getcpeid.so] => (Get ADSI CPE ID) [app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application) [app_zapateller.so] => (Block Tel...