search for: res_agi

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o [LD] res_clioriginate.o -> res_clioriginate.so [CC] res_convert.c -> res_convert.o [LD] res_convert.o -> res_convert.so [CC] res_crypto.c -> res_crypto.o [LD] r...

...ed=10") in new stack -- Executing SetVar("SIP/12-dec6", "authtype=sipextens") in new stack -- Executing AGI("SIP/12-dec6", "agi_cdr.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi_cdr.php Mar 24 05:06:18 WARNING[2911]: res_agi.c:376 printf_to_agi_result_buffer: Valid buffer! Mar 24 05:06:18 WARNING[2911]: res_agi.c:376 printf_to_agi_result_buffer: Valid buffer! Mar 24 05:06:18 WARNING[2911]: res_agi.c:376 printf_to_agi_result_buffer: Valid buffer! Mar 24 05:06:18 WARNING[2911]: res_agi.c:376 printf_to_agi_result_buffer:...

...create RTP audio session: Too many open files Feb 14 14:55:43 WARNING[11273] acl.c: Cannot create socket Feb 14 14:55:46 WARNING[11273] acl.c: Cannot create socket Feb 14 14:56:01 WARNING[11283] chan_misdn.c: Write returned <=0 (err=Destination address required) Feb 14 14:59:56 WARNING[13725] res_agi.c: unable to create fromast pipe: Too many open files Feb 14 15:01:21 WARNING[13729] res_agi.c: unable to create fromast pipe: Too many open files Feb 14 15:01:56 WARNING[13744] res_agi.c: Unable to create toast pipe: Too many open files Feb 14 15:01:56 ERROR[13744] cdr_custom.c: Unable to re-op...

...mand to > figure it out, but I can’t for the life of me find the underlying > source code. The closest I’ve found is XML that defines the command > syntax but can’t trace it from there. > Here is where it is in the master version: https://github.com/asterisk/asterisk/blob/master/res/res_agi.c#L2829 And in version 11: https://github.com/asterisk/asterisk/blob/11/res/res_agi.c#L2239

...nd get valid information. Sometimes "dialstatus" has the value "CONGESTION" OR "CANCEL". In this cases I tried to get the hangupcase with "GET FULL VARIABLE ${hangupcause}" but I always get an empty result. Examples: [...] [Jan 25 09:23:18] VERBOSE[28228] res_agi.c: <DAHDI/64-1>AGI Rx << GET FULL VARIABLE ${answeredtime} [Jan 25 09:23:18] VERBOSE[28228] res_agi.c: <DAHDI/64-1>AGI Tx >> 200 result=1 () [...] [Jan 25 09:51:00] VERBOSE[1349] res_agi.c: <DAHDI/33-1>AGI Rx << GET FULL VARIABLE ${dialstatus} [Jan 25 09:51:00...

Hi, I have a question about this here: Asterisk-Version: 11.10.2 File: res/res_agi.c Line: 2377 [...] static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[]) 2304 { 2305 struct ast_filestream *fs; 2306 struct ast_frame *f; 2307 struct timeval start; 2308 long sample_offset = 0; 2309 int res...

Hello everyone, I have a problem with a FastAGI connection, could you help me fix this problem please? Here is my log: [2015-12-15 16:17:09] WARNING[23936][C-00000015]: res_agi.c:1658 handle_connection: Connecting to '10.171.54.149:9110' failed for url 'agi://10.171.54.149:9110/DatabaseQuery.agi?vars=name%3Bfirst+name%3BStatut&db=Oracle+database&query=VIP&as=10.171.54.149': Connection refused [2015-12-15 16:17:09] WARNING[23936][C-00000015]: re...

...[root@box src]# gcc -v Reading specs from /usr/lib/gcc-lib/i386-redhat-linux/2.96/specs gcc version 2.96 20000731 (Red Hat Linux 7.2 2.96-112.7.2) [root@box src]# rpm -q unixODBC unixODBC-2.0.7-3 [root@box src]# rpm -q unixODBC-devel unixODBC-devel-2.0.7-3 ---beta1--- gcc -shared -Xlinker -x -o res_agi.so res_agi.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing- declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 - march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer - DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC -c -o res_odbc.o res_odbc.c res_od...

...risk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The command "asyncagi break" does stop ongoing playing but also breaks the async agi control. I only want the first. Thanks in advance, /Henrik

...OSE[1264031][C-00000025] pbx.c: Executing > [s at function-smartscreen:2] EAGI("PJSIP/Twilio-NA-W-3-In-00000068", > "smartscreen/smartscreen.php,"GEORGE SMITH" <+1234567890>") in new > stack > > [2023-06-30 15:50:47] VERBOSE[1264031][C-00000025] res_agi.c: Launched > AGI Script /var/lib/asterisk/agi-bin/smartscreen/smartscreen.php > > [2023-06-30 15:50:48] VERBOSE[1264031][C-00000025] res_agi.c: > <PJSIP/Twilio-NA-W-3-In-00000068>AGI Script > smartscreen/smartscreen.php completed, returning 0 > > I never see any mes...

...;s,1,Answer() exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)}) exten=>s,3,Goto(check_time,s,1) and the FastAGI server is not running (Asterisk gets "connection refused" TCP error), Asterisk just terminates the call like so: May 17 12:58:00 WARNING[10154]: res_agi.c:210 launch_netscript: Connect to 'agi://192.168.1.175/calldirect?check&NNNNNNNNNN' failed: Connection refused == Spawn extension (remove_caller, s, 2) exited non-zero on 'SIP/datatrak-0978b670' The text of the relevant wiki article (http://www.voip-info.org/wiki-Asteri...

...var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC -c -o res_agi.o res_agi.c gcc -shared -Xlinker -x -o res_agi.so res_agi.o make[1]: Leaving directory `/usr/local/src/development/asteriskv1-0_1.0.9/asterisk/res' make[1]: Entering directory `/usr/local/src/development/asteriskv1-0_1.0.9/asterisk/channels' gcc -pipe -Wall -Wstrict-prototypes -Wmissin...

Hi all, Does anybody know why my asterisk doesn't have a "show agi" command? Do I have to load any module for it? Thanks Ronaldo.

Hi I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my DeadAGI scripts are not working properly. Like after hangup I used to do some more work now its not working. Please help. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070722/09df42e9/attachment.htm

...39;app_voicemail.so': /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: ast_smdi_mwi_message_destroy loader.c: Error loading module 'res_fax_spandsp.so': /usr/lib/asterisk/modules/res_fax_spandsp.so: undefined symbol: ast_fax_tech_register loader.c: Error loading module 'res_agi.so': /usr/lib/asterisk/modules/res_agi.so: undefined symbol: ast_speech_start loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client loader.c: Error loading module 'chan_mgcp.so': /usr/lib/asterisk/modules/ch...

...led: Can't create alert pipe! Apr 5 00:48:36 WARNING[14887]: chan_local.c:523 local_new: Unable to allocate channel structure(s) Apr 5 08:48:36 NOTICE[14887]: app_dial.c:1042 dial_exec_full: Unable to create channel of type 'LOCAL' (cause 0 - Unknown) Apr 5 08:48:36 WARNING[14893]: res_agi.c:246 launch_script: unable to create fromast pipe: Too many open files Apr 5 08:48:37 WARNING[14894]: res_agi.c:246 launch_script: unable to create fromast pipe: Too many open files Apr 5 08:48:38 WARNING[14897]: channel.c:562 ast_channel_alloc: Channel allocation failed: Can't create ale...

...gt; ASTERISK_FILE_VERSION(__FILE__, "$Revision: 59360 $") 1299,1302d1298 < /* If the channel wants to be hung up, hang it up */ < if (ast_check_hangup(chan)) < break; < And here is the entire diff from res_agi.c: [root at asterisk asterisk]# diff res/res_agi.c /usr/src/asterisk-1.2.19/res/res_agi.c 44c44 < ASTERISK_FILE_VERSION(__FILE__, "$Revision: 71656 $") --- > ASTERISK_FILE_VERSION(__FILE__, "$Revision: 54771 $") 572c572,579 < ast_playstream(fs); --- >...

...in new stack [2011-03-13 09:51:54] VERBOSE[4699] pbx.c: -- Goto (a2billing,37129238254,1) [2011-03-13 09:51:54] VERBOSE[4699] pbx.c: -- Executing [37129238254 at a2billing:1] AGI("SIP/0010333-00000069", "a2billing.php,8") in new stack [2011-03-13 09:51:54] VERBOSE[4699] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php [2011-03-13 09:51:55] VERBOSE[4699] res_agi.c: -- AGI Script Executing Application: (DIAL) Options: (SIP/pozitel/37129238254,60,tTL(34980000:61000:30000)) [2011-03-13 09:51:55] VERBOSE[4699] netsock.c: == Using UDPTL TOS bi...

...-- Executing [100 at sip:1] AGI("SIP/userid-0000001e", "test.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script Executing Application: (SetCallerPres) Options: (prohib_not_screened) [Jan 24 15:20:04] WARNING[15507][C-00000030]: res_agi.c:2527 handle_exec: Could not find application (SetCallerPres) Why is the application not found, please? I think it should exist: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_SetCallerPres Best regards, -Thorsten-

...information leads you to suspect the ODBC drivers...? What I see in the log is: ================================================================ [Jun 7 14:23:58] VERBOSE[11347][C-00000001] app_dial.c: Everyone is busy/congested at this time (1:0/0/1) [Jun 7 14:23:58] VERBOSE[11347][C-00000001] res_agi.c: magic_switch.pl: --------------------------- jmd (CHANUNAVAIL) [Jun 7 14:23:58] VERBOSE[11347][C-00000001] res_agi.c: AGI Script Executing Application: (voicemail) Options: (1505903xxxx at default,su) [Jun 7 14:23:58] VERBOSE[11347][C-00000001] file.c: <SIP/FFFF548xxxxx-1-00000000> Playi...