Re: [Asterisk-Users] Is there a right place for a include_oncestatement in a PHP AGI script?
agree with all written below - additionally use php -l to lint/check the syntax of the file (and the include) if needed - do a include_once 'bleh.php || die "some message"; to see if thats an issue. my $0.02 - Original Message - From: "Moises Silva" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, August 05, 2005 3:49 PM Subject: Re: [Asterisk-Users] Is there a right place for a include_oncestatement in a PHP AGI script? its kind of difficult to say if we dont know what the included php script has. i think that the wrap function that Christoph propouse it may work for debuggin purposes, but i dont think it will solve the problem. Until you tell us, or show us, the content of the scripts we will be doing our best to guess the problem. I think you have parse error in the included script, try turning on the log errors directives in php.ini, turn off the output errors stuff, so Asterisk will not get confused with php warnings and other stuff. Let us know what happen... best regards On 8/5/05, Christoph Eicke <[EMAIL PROTECTED]> wrote: On Friday 05 August 2005 14:04, Leo Burd wrote: > Hello there, > > I'm new to PHP AGIs and I'm having problems with a particular script > that has a "include_once" statement on it. If I remove that stament, > the script runs until the section of the code that depends on the > include and then returns. If I include that statement, the script does > not seem to run at all. What shall I do? Leo, wrap a function around whatever is in the included script, make your include_once() statement at the top of the AGI and then simply call the function at the place where it's necessary for that code to be executed. Christoph ___ 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 -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; ___ 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
Re: ARTCP Project (was RE: [Asterisk-Users] Features you'd like to see ina GUI?)
Outta intersted - why mysql? If postgresql not a better option? I would happily contribute to postgres work (and am indeed starting to work on something similar atm, schemas written, etc) - but at the end of the day mysql still does not cut it inmo. No offence to mysql developers, etc. Cheers Chris - Original Message - From: "Sherwood McGowan" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Friday, August 05, 2005 2:06 PM Subject: ARTCP Project (was RE: [Asterisk-Users] Features you'd like to see ina GUI?) I'd like to officially "reclaim" the Features in a GUI thread ;) Asterisk Hackers, Admins, and general digital phreakers of old, After careful consideration, the ARTCP project will probably have to be split into two major sections, both distros or at least maps for a system to be designed as well as the software we develop. Section 1. ARTCP Provider This version will be a distro including the designed management/enduser/billing software hooking into an asterisk RT installation using pre-set mysql schemas. The reasoning for this is that it's much easier to design a database driven php package when you know what the schema will be. Section 2. ARTCP PBX Intended for medium to large pbx's for endusers that want RealTime performance, this project will be a distro with less overall features, but more than enough to handle PBX functions and more. Possible Section 3? AI-PBX Cpanel? (name?) This version will be the same as ARTCP PBX, but not running the RT version of Asterisk. All the above sections should have: A complete branded distribution of linux, as small as possible. Bacula backup system Zabbix monitoring system Asterisk (STABLE) All Asterisk apps/modules that are required for final product PHP MySQL Apache Samba (for end user uploading music on hold files) Webmin (for end user control of system) I'd like to try and get anyone interested in contributing code work to join me in an online IRC chatroom sometime around August 17, 2005. Please reply (just please erase the [Asterisk-Users] section of your subject, otherwise it'll get trapped in the general list folder due to message moving rules) to me directly, and we'll get everyone's availability worked out. I'll take contributions of funds as well to be split across the developers, but won't look for donations before at least some cursory info has been released to show that the project is at least happening ;) Cheers all, and I hope to see interest in getting this going. Sherwood McGowan ___ 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
Re: [Asterisk-Users] Need Advice
Lucky you - i have had 2 die before i opted for the TDM. - Original Message - From: "Nathan Pralle" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, July 25, 2005 2:40 PM Subject: Re: [Asterisk-Users] Need Advice ACK! I forgot about that. You're right, my bad. Still, they seem to work well. Nathan Eric Wieling aka ManxPower wrote: Nathan Pralle wrote: However, for FXO ports, I'm using the Digium Wildcard X100P's which can be obtained on eBay for $9-$20, usually. Much cheaper price-per-port, although the TDM would give better expandibility. You mean NON Digium X100P's. Digium no longer sells the X100P. The cheap ones on eBay are "clone" cards. -- - Nathan E. Pralle Give the director a serpent deflector. www.nathanpralle.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
Re: [Asterisk-Users] Uk Caller id
Caller id is deffinately passed (checked by plugging phone straight into line). So its me + asterisk is the problem :) most likely my config. Cheers Chris - Original Message - From: "Giorgio Incantalupo" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, July 25, 2005 10:53 AM Subject: Re: [Asterisk-Users] Uk Caller id Hi, we had 3 analog lines but our telco did't pass us the caller-id so Asterisk tried to identify the caller-id but found bad data. Setting usecallerid=no avoided the Warning. Be sure your telco sends you the caller id. Giorgio Chris Thompson wrote: Hey Thanks for the response but still no luck. I'm a bit baffeled, i might try running some of the old versions + patch and stick in my old x100p cards to see what happens unless anyone has any great ideas? However I didn't really understand why i would use usecallerid=no when i do* want to use caller id. Cheers Chris - Original Message - From: "Giorgio Incantalupo" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, July 22, 2005 4:17 PM Subject: Re: [Asterisk-Users] Uk Caller id Hi, put usecallerid=no Giorgio Chris Thompson wrote: Hi I have my new TDM400P installed and working. I'm running from cvs HEAD with a 2.6.12 kernel on debian. I can't seem to get Caller id working (in uk with clid supplied and working to line) but am a bit unclear on the docs and hence assume it is something I am doing wrong. I would really* appreciate if anyone could take a look below at my zapata.conf and see is there anything incorrect. I am least convinced on the usecallerid=uk option, but if set to 'yes' i get Jul 22 15:38:47 ERROR[19569]: callerid.c:266 callerid_feed: fsk_serie made mylen < 0 (-20) Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5796 ss_thread: CallerID feed failed: Success Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5840 ss_thread: CallerID returned with error on channel 'Zap/2-1' :: zapata.conf :: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=uk callerid=asreceived cidsignalling=v23 cidstart=usehist callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 immediate=no progzone=uk musiconhold=default ; incoming channels signalling=fxs_ks group=2 context=incoming channel => 1-2 ; outgoing channels signalling=fxo_ks group=1 context=outgoing channel => 3 Thanks loads Chris ___ 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 -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.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
Re: [Asterisk-Users] Uk Caller id
Hey Thanks for the response but still no luck. I'm a bit baffeled, i might try running some of the old versions + patch and stick in my old x100p cards to see what happens unless anyone has any great ideas? However I didn't really understand why i would use usecallerid=no when i do* want to use caller id. Cheers Chris - Original Message - From: "Giorgio Incantalupo" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, July 22, 2005 4:17 PM Subject: Re: [Asterisk-Users] Uk Caller id Hi, put usecallerid=no Giorgio Chris Thompson wrote: Hi I have my new TDM400P installed and working. I'm running from cvs HEAD with a 2.6.12 kernel on debian. I can't seem to get Caller id working (in uk with clid supplied and working to line) but am a bit unclear on the docs and hence assume it is something I am doing wrong. I would really* appreciate if anyone could take a look below at my zapata.conf and see is there anything incorrect. I am least convinced on the usecallerid=uk option, but if set to 'yes' i get Jul 22 15:38:47 ERROR[19569]: callerid.c:266 callerid_feed: fsk_serie made mylen < 0 (-20) Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5796 ss_thread: CallerID feed failed: Success Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5840 ss_thread: CallerID returned with error on channel 'Zap/2-1' :: zapata.conf :: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=uk callerid=asreceived cidsignalling=v23 cidstart=usehist callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 immediate=no progzone=uk musiconhold=default ; incoming channels signalling=fxs_ks group=2 context=incoming channel => 1-2 ; outgoing channels signalling=fxo_ks group=1 context=outgoing channel => 3 Thanks loads Chris ___ 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
[Asterisk-Users] Uk Caller id
Hi I have my new TDM400P installed and working. I'm running from cvs HEAD with a 2.6.12 kernel on debian. I can't seem to get Caller id working (in uk with clid supplied and working to line) but am a bit unclear on the docs and hence assume it is something I am doing wrong. I would really* appreciate if anyone could take a look below at my zapata.conf and see is there anything incorrect. I am least convinced on the usecallerid=uk option, but if set to 'yes' i get Jul 22 15:38:47 ERROR[19569]: callerid.c:266 callerid_feed: fsk_serie made mylen < 0 (-20)Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5796 ss_thread: CallerID feed failed: SuccessJul 22 15:38:47 WARNING[19569]: chan_zap.c:5840 ss_thread: CallerID returned with error on channel 'Zap/2-1' :: zapata.conf :: [channels]context=defaultswitchtype=nationalsignalling=fxo_lsrxwink=300 ; Atlas seems to use long (250ms) winksusecallerid=ukcallerid=asreceivedcidsignalling=v23cidstart=usehistcallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesgroup=1callgroup=1pickupgroup=1immediate=noprogzone=ukmusiconhold=default ; incoming channels signalling=fxs_ksgroup=2context=incomingchannel => 1-2 ; outgoing channels signalling=fxo_ksgroup=1context=outgoingchannel => 3 Thanks loads Chris ___ 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
