[Asterisk-Users] Re: Problem with call hold
FYI, I just download the latest stable version from CVS and the problem is gone. - Original Message - From: Joseph Shi To: asterisk-users@lists.digium.com Sent: Monday, February 28, 2005 9:22 PM Subject: Problem with call hold I got a very strange problem with call-hold function. For calls that come in from PSTN and route to a SIP extension. If I put the call on hold, I cannot unhold the call after. The caller would be left with hold music forever. A warning message would be shown on the console usually a few seconds after putting the call on hold: WARNING[17428]: chan_sip.c:686 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2079 (non-critical Response). The same unhold function works fine for calls between SIP extensions. I have searched through wiki but could not find the answer. If somebody can shred some light on the problem, it will be very much appreciated. I'm running the Asterisk stable version at Dec 21, 2004. Thanks ahead. Joseph Shi ___ 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] Addons Make Problems! HELP!
On Wed, Mar 02, 2005 at 08:45:24AM +0100, Daniele Gallina - 3P System S.r.l. wrote: Hi all, I have a server with an Athlon 64 3200 and Fedora Core 2 x86_64. I have compiled and installed Asterisk 1.0.6 without any problems. When I try to make asterisk-addons-1.0.6 it say me: [EMAIL PROTECTED] asterisk-addons-1.0.6]# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons-1.0.6/format_mp3' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o common.o common.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o dct64_i386.o dct64_i386.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o decode_ntom.o decode_ntom.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o layer3.o layer3.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o tabinit.o tabinit.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o interface.o interface.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o format_mp3.o format_mp3.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o /usr/bin/ld: common.o: relocation R_X86_64_32 can not be used when making a shared object; recompile with -fPIC Not that I really know what this means, but any chance you're mixing 32bit and 64bit code? common.o: could not read symbols: Bad value collect2: ld returned 1 exit status make[1]: *** [format_mp3.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk-addons-1.0.6/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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] Forward Call from Asterisk to SER
I have some problem to redirect the call from asterisk to ser. 1 thing i am redirecting call to asterisk and then on some extension i want to return the call to ser. Receiving this error: WARNING[23594]: chan_sip.c:6829 handle_response: Forbidden - wrong password on authentication for INVITE to '"Alex" sip:[EMAIL PROTECTED]:5061;tag=as55a3adbb' -- SIP/212.25.75.195:5060-3bc0 is circuit-busy Any help will be appreciate. Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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] Asterisk realtime , asterisk extensions not load form db.
Hello ! I try to run asterisk with real time config from database. I use AMP to configure . Everythig it ok , I cansetnew sip and iax extensions, I can see them on mysql db , as well is amp . But these extension I cannt use in asterisk . I have seen some new conf file ;sip_additional.conf;iax_additional.conf;extensions_additional.conf;meetme_additional.conf .If I reload asterisk these extensions are not load . I have these tabele in mysql ,sip, iax, extensions. my extconfig.conf file [settings] ;uncomment to load queues.conf via the db engine.;queues.conf = odbc;conf filename = driver,databasename[,table_name] voicemail.conf = mysql,asterisk,voicemail_table; it is goodsip.conf = mysql,asterisk,sip_tableiax = mysql,asterisk,iaxextensions.conf = mysql,asterisk,extensions How can I set asterisk to load config form database ? Is there anyone who have the same problem ? Thans for help ! Roby. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Re: Dial application invoked again and again
hi If i remove _. from my dialplan(extensions.conf). application is invoked only once. otherwise application is invoked again and again. any one know what is the problem and how to make (global) dialplan for all user agents. thanks Kamran __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ 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] [OT] stupid firmware question...
On 2 Mar 2005, at 20:24, Chris Wade wrote: I know this is a really stupid question, but I just have to ask... Where would I start if I wanted to try and develop my own firmware for a particular phone. Namely, I want to try and 're-write' the SIP firmware for Cisco 7940's. Any ideas? -Chris PS: [* put on flame suit *] why won't any of the phone manufacturer's just open-source the firmware for their phones? [* ducks head back inside gopher hole just before a giant fireball hits *] Snom do. At least there is this link on their website http://www.snom.com/snom_source.html?L=1 I've not used the source as yet, but it was one of the factors that persuaded me to buy phones from them. Tim. http://www.westhawk.co.uk/ ___ 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] dropping extra frame..already have it????
Recently, I've been getting these messages: Mar 1 13:59:44 NOTICE[20933]: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Well I got the same when I started to use g729. I did some search crawl in the archive and fount in the 'dev' part two mails about that. It seems g729 on asterisk does not like variable length packets. Disabling silence compression on the UA did help me. I did also upgrade the UA's bios at the same time. I am an asterisk newbie, so I cannot guarantee the answer. Give some feed-back. Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail shaoul (at) tellink.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] Getting phpconfig to work?
Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description Parent Directory03-Mar-2005 12:15 - asterisk.reload 03-Mar-2005 12:28 1k cls_phpconfig.php 03-Mar-2005 11:4814k cls_phpconfig_html.php 03-Mar-2005 11:5517k images/ 24-Feb-2005 09:06 - phpconfig.php 14-Sep-2003 19:32 6k phpconfig_init.php 03-Mar-2005 11:44 2k Apache/1.3.33 Server at ip-of-machine Port 80 I have made the necessary changes to all the files in the phpconfig directory and my DocumentRoot is set to /usr/local/www/. To add to this, I have php4-4.3.10_1 and apache+mod_ssl-1.3.33+2.8.22 installed on my Asterisk box. What could I be doing wrong? Thanks in advance! Rgds, Julius. ___ 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] Multitenant feature
Hi all, Has any one tested or know if Asterisk support multitenant PBX, ie the Asterisk support either multiinstances on the same machines or acts exactly as many virtual PBXs to be shared between several small campanies. Thanks for the hint Aref ___ 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] Getting phpconfig to work?
Try using the url http://ip-of-machine/phpconfig/phpconfig.php On Thu, 3 Mar 2005 12:45:03 +0300 (EAT), Julius Kidubuka [EMAIL PROTECTED] wrote: Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description Parent Directory03-Mar-2005 12:15 - asterisk.reload 03-Mar-2005 12:28 1k cls_phpconfig.php 03-Mar-2005 11:4814k cls_phpconfig_html.php 03-Mar-2005 11:5517k images/ 24-Feb-2005 09:06 - phpconfig.php 14-Sep-2003 19:32 6k phpconfig_init.php 03-Mar-2005 11:44 2k Apache/1.3.33 Server at ip-of-machine Port 80 I have made the necessary changes to all the files in the phpconfig directory and my DocumentRoot is set to /usr/local/www/. To add to this, I have php4-4.3.10_1 and apache+mod_ssl-1.3.33+2.8.22 installed on my Asterisk box. What could I be doing wrong? Thanks in advance! Rgds, Julius. ___ 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] Multitenant feature
Contexts can be used to partition Asterisk, but the administration is not multitenanted On Thu, 3 Mar 2005 10:47:03 +0100, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, Has any one tested or know if Asterisk support multitenant PBX, ie the Asterisk support either multiinstances on the same machines or acts exactly as many virtual PBXs to be shared between several small campanies. Thanks for the hint Aref ___ 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] Asterisk-OH323 no ringing
I tried every possibility of H245/Faststart ... don't change anything. It seems that if I do a call IAX-IAX the ringtone is here, but not if doing H323-IAX or IAX-H323. If I can't find the problem, I will submit to your bugtracker, but I thought asterisk-oh323 is quite popular and I'm sure I'm not the first using it this way. So why would I be the first to have this problem ? Or am I the only one who need a ringtone ? :) I would be glad to send more infos but - don't know what could really be usefull because I have no idea what is the problem here. For everyone on the mailing list ... if you had a similar problem please tell me how you solved it. Or if you have any idea what I could try. Thankyou. George K. Konstantoulakis wrote: Hello Yves, please try switching off fastStart, if the problem persists please submit it to our bugtracker : https://skylab.inaccessnetworks.com/mantis and we will do our best to help you. George. Yves wrote: First thank you for answering. Ok, I'll try to be as clear and precise as possible : I have misc. incoming calls (SIP H323), when a call arrive I don't answer in the dialplan. Every call is going through a perl script (via asterisk-perl) who handle the routing process. Depending a lot of parameters, I choose which trunk I will use for outgoing (this part is working well). Then I just use a dial command (in the perl script) to process the call out. When it's a SIP-IAX, IAX-IAX, ... everything is fine. But when I call arrive on my PBX in H323, the customers conplain that there is no ring tone while the phone is ringing on the other side. Just a long silence ... and when the called person pickup the phone, everything is just right. ( I reproduced this with gnomemeetingnetmeeting). I also get this notice in the asterisk console : NOTICE[4242]: chan_oh323.c:1366 oh323_indicate: Ignoring PROGRESS indication. Maybe it has something to do with it. oh323.conf : default config, except I have faststart on If you need more precise about something (config or what) just tell me, I don't really know what more I could tell you. George K. Konstantoulakis wrote: Hello Yves, could you please describe in more detail your problem. If you Answer() the call in the dialplan it is the correct behaviour not to hear any ringinging. If this is not the case please supply more information about your setup so that we can help you. George. Yves wrote: Hello, I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5). Everything is working fine, well, except that : when a call is made from an h323 device (gnomemeeting for example), the caller does not hear any ringing at all, he suddenly hears the person who answers the phone. That can be quite disturbing for the users. Any help would be very welcome. thank you. Yves ___ 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] Asterisk realtime , asterisk extensions not load form db.
Title: Message You've got to check if you have all the required mysql libraries installed (mysql client and mysql-devel) -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk asteriskSent: Donnerstag, 3. März 2005 10:13To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk realtime ,asterisk extensions not load form db. Hello ! I try to run asterisk with real time config from database. I use AMP to configure . Everythig it ok , I cansetnew sip and iax extensions, I can see them on mysql db , as well is amp . But these extension I cannt use in asterisk . I have seen some new conf file ;sip_additional.conf;iax_additional.conf;extensions_additional.conf;meetme_additional.conf .If I reload asterisk these extensions are not load . I have these tabele in mysql ,sip, iax, extensions. my extconfig.conf file [settings] ;uncomment to load queues.conf via the db engine.;queues.conf = odbc;conf filename = driver,databasename[,table_name] voicemail.conf = mysql,asterisk,voicemail_table; it is goodsip.conf = mysql,asterisk,sip_tableiax = mysql,asterisk,iaxextensions.conf = mysql,asterisk,extensions How can I set asterisk to load config form database ? Is there anyone who have the same problem ? Thans for help ! Roby. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] RE: Getting phpconfig to work?
Hi, When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; ?PHP /** * * Asterisk configuration file interface script * * * * * * phpconfig:,v 1.0 2003/07/03 17:19:37 * Authors: Dave Packham [EMAIL PROTECTED] * Rob Birkinshaw [EMAIL PROTECTED] */ require_once(phpconfig_init.php); require_once(cls_phpconfig.php); require_once(cls_phpconfig_html.php); // create and initalize objects // $page = new Open_Conf_HTML(); $conf = new Open_Conf(); // setup 3pane html theme $page-OC_HTML_set3Pane(true); // pass thru phpconfig_init paramaters to object $conf-OC_setConfDirectories($conf_directories); $conf-OC_setTempDir($temporary_directory); $conf-OC_setTempPrefix($temporary_file_prefix); $conf-OC_setAccessFile($access_file); $conf-OC_setConfDir($default_conf_file_direcotry,$conf_directories); $conf-OC_setConfFilter($conf_file_filter); $conf-OC_setResetCmd($reset_cmd); $conf-OC_setRemark($remark); // pass thru html output parameters $page-OC_HTML_setTextareaRows($textarea_rows); $page-OC_HTML_setImagesDir($images_dir); $page-OC_HTML_setLogo($logo); $page-OC_HTML_setTitle($title); $page-OC_HTML_setPageTitle($page_title); $page-OC_HTML_setDescription($description); $page-OC_HTML_setKeywords($keywords); $page-OC_HTML_setWebmaster($webmaster); $page-OC_HTML_setDisclaimer($disclaimer); $page-OC_HTML_setFooterText($footer_text); $page-OC_HTML_setLogoLink($logo_link); // init top menu bar $page-OC_HTML_setHeaderBar(array_merge($conf-_OC_conf_dirs , array(Re-Read Configs=phpconfig.php?reset=reset))); // init side menus $menuList = array(); $rightMenuList = array(); session_start(); // this session variable will be set by a login screen // in a future release // for now, fake it for the prototype $_SESSION['valid_user'] = $fakeuser; $conf-OC_checkValidUser(); if($_GET['file']) // conf file requested via menu How do I proceed from here? Thanks, Julius. Hi, You need to click on the phpconfig.php to actually load the page. It is because the file is not called index.php. This could be changed in the apache conf file if wished. Let us know how you go. C -Original Message- From: World Wide Web Owner [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 09:45 To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: Getting phpconfig to work? Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description Parent Directory03-Mar-2005 12:15 - asterisk.reload 03-Mar-2005 12:28 1k cls_phpconfig.php 03-Mar-2005 11:4814k cls_phpconfig_html.php 03-Mar-2005 11:5517k images/ 24-Feb-2005 09:06 - phpconfig.php 14-Sep-2003 19:32 6k phpconfig_init.php 03-Mar-2005 11:44 2k Apache/1.3.33 Server at ip-of-machine Port 80 I have made the necessary changes to all the files in the phpconfig directory and my DocumentRoot is set to /usr/local/www/. To add to this, I have php4-4.3.10_1 and apache+mod_ssl-1.3.33+2.8.22 installed on my Asterisk box. What could I be doing wrong? Thanks in advance! Rgds, Julius. -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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] RE: Getting phpconfig to work?
Hi, When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; ?PHP /** * * Asterisk configuration file interface script * * * * * * phpconfig:,v 1.0 2003/07/03 17:19:37 * Authors: Dave Packham [EMAIL PROTECTED] * Rob Birkinshaw [EMAIL PROTECTED] */ require_once(phpconfig_init.php); require_once(cls_phpconfig.php); require_once(cls_phpconfig_html.php); // create and initalize objects // $page = new Open_Conf_HTML(); $conf = new Open_Conf(); // setup 3pane html theme $page-OC_HTML_set3Pane(true); // pass thru phpconfig_init paramaters to object $conf-OC_setConfDirectories($conf_directories); $conf-OC_setTempDir($temporary_directory); $conf-OC_setTempPrefix($temporary_file_prefix); $conf-OC_setAccessFile($access_file); $conf-OC_setConfDir($default_conf_file_direcotry,$conf_directories); $conf-OC_setConfFilter($conf_file_filter); $conf-OC_setResetCmd($reset_cmd); $conf-OC_setRemark($remark); // pass thru html output parameters $page-OC_HTML_setTextareaRows($textarea_rows); $page-OC_HTML_setImagesDir($images_dir); $page-OC_HTML_setLogo($logo); $page-OC_HTML_setTitle($title); $page-OC_HTML_setPageTitle($page_title); $page-OC_HTML_setDescription($description); $page-OC_HTML_setKeywords($keywords); $page-OC_HTML_setWebmaster($webmaster); $page-OC_HTML_setDisclaimer($disclaimer); $page-OC_HTML_setFooterText($footer_text); $page-OC_HTML_setLogoLink($logo_link); // init top menu bar $page-OC_HTML_setHeaderBar(array_merge($conf-_OC_conf_dirs , array(Re-Read Configs=phpconfig.php?reset=reset))); // init side menus $menuList = array(); $rightMenuList = array(); session_start(); // this session variable will be set by a login screen // in a future release // for now, fake it for the prototype $_SESSION['valid_user'] = $fakeuser; $conf-OC_checkValidUser(); if($_GET['file']) // conf file requested via menu How do I proceed from here? Thanks, Julius. Hi, You need to click on the phpconfig.php to actually load the page. It is because the file is not called index.php. This could be changed in the apache conf file if wished. Let us know how you go. C -Original Message- From: World Wide Web Owner [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 09:45 To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: Getting phpconfig to work? Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description Parent Directory03-Mar-2005 12:15 - asterisk.reload 03-Mar-2005 12:28 1k cls_phpconfig.php 03-Mar-2005 11:4814k cls_phpconfig_html.php 03-Mar-2005 11:5517k images/ 24-Feb-2005 09:06 - phpconfig.php 14-Sep-2003 19:32 6k phpconfig_init.php 03-Mar-2005 11:44 2k Apache/1.3.33 Server at ip-of-machine Port 80 I have made the necessary changes to all the files in the phpconfig directory and my DocumentRoot is set to /usr/local/www/. To add to this, I have php4-4.3.10_1 and apache+mod_ssl-1.3.33+2.8.22 installed on my Asterisk box. What could I be doing wrong? Thanks in advance! Rgds, Julius. ___ 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] Getting phpconfig to work?
That url returns the actual contents of the phpconfig file and doesn't load the page as required. How can I go about it? Thanks, Julius. Try using the url http://ip-of-machine/phpconfig/phpconfig.php On Thu, 3 Mar 2005 12:45:03 +0300 (EAT), Julius Kidubuka [EMAIL PROTECTED] wrote: Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description Parent Directory03-Mar-2005 12:15 - asterisk.reload 03-Mar-2005 12:28 1k cls_phpconfig.php 03-Mar-2005 11:4814k cls_phpconfig_html.php 03-Mar-2005 11:5517k images/ 24-Feb-2005 09:06 - phpconfig.php 14-Sep-2003 19:32 6k phpconfig_init.php 03-Mar-2005 11:44 2k Apache/1.3.33 Server at ip-of-machine Port 80 I have made the necessary changes to all the files in the phpconfig directory and my DocumentRoot is set to /usr/local/www/. To add to this, I have php4-4.3.10_1 and apache+mod_ssl-1.3.33+2.8.22 installed on my Asterisk box. What could I be doing wrong? Thanks in advance! Rgds, Julius. ___ 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] RE: Getting phpconfig to work?
Hi, Does phpconfig require a particular php package installed? I have php4-4.3.10_1 installed on my box. Does this have an effect? What do I need to change in terms of ownership and permissions to files located in the phpconfig directory? At the moment I have, drwxr-xr-x 4 root wheel512 Mar 3 13:09 . drwxr-xr-x 8 root wheel512 Mar 3 12:15 .. drwxr-xr-x 2 root wheel512 Feb 24 09:06 CVS -rwxrwxrwx 1 root wheel588 Mar 3 12:28 asterisk.reload -rw-r--r-- 1 root wheel 14702 Mar 3 11:48 cls_phpconfig.php -rw-r--r-- 1 root wheel 17846 Mar 3 11:55 cls_phpconfig_html.php drwxr-xr-x 3 root wheel512 Feb 24 09:06 images -rw-r--r-- 1 root wheel 6515 Sep 14 2003 phpconfig.php -rw-r--r-- 1 root wheel 2128 Mar 3 11:44 phpconfig_init.php Any indicators? Rgds, Julius. That means PHP isn't setup properly on your server; it isn't parsing the file as PHP. It is just outputting the code. Create a file named test.php and put this in the file: ?php phpinfo(); ? This should display lots of information in a table about your php installation. When you have this working; phpconfig should work. C -Original Message- From: World Wide Web Owner [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 10:02 To: C. Tomlinson Subject: RE: Getting phpconfig to work? Hi, When I do click on the phpconfig.php link, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output from http://ip-of-machine/phpconfig/phpconfig.php below; ?PHP /** * * Asterisk configuration file interface script * * * * * * phpconfig:,v 1.0 2003/07/03 17:19:37 * Authors: Dave Packham [EMAIL PROTECTED] * Rob Birkinshaw [EMAIL PROTECTED] */ require_once(phpconfig_init.php); require_once(cls_phpconfig.php); require_once(cls_phpconfig_html.php); // create and initalize objects // $page = new Open_Conf_HTML(); $conf = new Open_Conf(); // setup 3pane html theme $page-OC_HTML_set3Pane(true); // pass thru phpconfig_init paramaters to object $conf-OC_setConfDirectories($conf_directories); $conf-OC_setTempDir($temporary_directory); $conf-OC_setTempPrefix($temporary_file_prefix); $conf-OC_setAccessFile($access_file); $conf-OC_setConfDir($default_conf_file_direcotry,$conf_directories); $conf-OC_setConfFilter($conf_file_filter); $conf-OC_setResetCmd($reset_cmd); $conf-OC_setRemark($remark); // pass thru html output parameters $page-OC_HTML_setTextareaRows($textarea_rows); $page-OC_HTML_setImagesDir($images_dir); $page-OC_HTML_setLogo($logo); $page-OC_HTML_setTitle($title); $page-OC_HTML_setPageTitle($page_title); $page-OC_HTML_setDescription($description); $page-OC_HTML_setKeywords($keywords); $page-OC_HTML_setWebmaster($webmaster); $page-OC_HTML_setDisclaimer($disclaimer); $page-OC_HTML_setFooterText($footer_text); $page-OC_HTML_setLogoLink($logo_link); // init top menu bar $page-OC_HTML_setHeaderBar(array_merge($conf-_OC_conf_dirs , array(Re-Read Configs=phpconfig.php?reset=reset))); // init side menus $menuList = array(); $rightMenuList = array(); session_start(); // this session variable will be set by a login screen // in a future release // for now, fake it for the prototype $_SESSION['valid_user'] = $fakeuser; $conf-OC_checkValidUser(); if($_GET['file']) // conf file requested via menu How do I proceed from here? Thanks, Julius. Hi, You need to click on the phpconfig.php to actually load the page. It is because the file is not called index.php. This could be changed in the apache conf file if wished. Let us know how you go. C -Original Message- From: World Wide Web Owner [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 09:45 To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: Getting phpconfig to work? Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description Parent Directory03-Mar-2005 12:15 - asterisk.reload 03-Mar-2005 12:28 1k cls_phpconfig.php 03-Mar-2005 11:4814k cls_phpconfig_html.php 03-Mar-2005 11:5517k images/ 24-Feb-2005 09:06 - phpconfig.php 14-Sep-2003 19:32 6k phpconfig_init.php 03-Mar-2005 11:44 2k Apache/1.3.33 Server at ip-of-machine Port 80 I have made the necessary
Re: [Asterisk-Users] Multitenant feature
Thnaks Aref Selon Jason Williams [EMAIL PROTECTED]: Contexts can be used to partition Asterisk, but the administration is not multitenanted On Thu, 3 Mar 2005 10:47:03 +0100, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, Has any one tested or know if Asterisk support multitenant PBX, ie the Asterisk support either multiinstances on the same machines or acts exactly as many virtual PBXs to be shared between several small campanies. Thanks for the hint Aref ___ 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] RE: Getting phpconfig to work?
When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; Try a simple php-script in this directory. Something like this, name it test.php ?PHP print (test) ? If it does not print test, and you see the code instead, check your httpd.conf for a general php-parsing problem. Regards Guido Hecken ___ 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] Realtime IAX/SIP with 2 asterisk servers but 1 central iax/sipfriends Database
Hello I was wandering If I let 2 asterisk boxes (let's name them ast01 and ast02) connect to one SQL realtime iaxfriends/sipfriends database What happens if I register my client to ast01, The ast01 box will update the client's record in the iaxfriends database (ipaddr/port/regseconds) Let's say there is an incoming call then for this client but this call arrives on ast02 (the box where the client is NOT registerd to at the moment) .. ast02 will 'know' then (with a DB lookup in the same table) where (which ipaddr/port) to route the call to am I right? will that work? And... Will that work too if the client is behind a NAT? ___ 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] Asterisk realtime , asterisk extensions not load form db.
Yes I've checked . these pakeche I have instaled. But it does not work. echo "libxml2"rpm -qa|grep libxml2echo "libtiff"rpm -qa|grep libtiffecho "libtiff-devel"rpm -qa|grep libtiff-develecho "httpd"rpm -qa|grep httpdecho "mysql"rpm -qa|grep mysqlecho "mysql -devel"rpm -qa|grep mysql-develecho "mysql -server"rpm -qa|grep mysql-serverecho "php"rpm -qa|grep phpecho "php -mysql"rpm -qa|grep php-mysqlecho "openssl"rpm -qa|grep opensslecho "openssl -devel" sp;rpm -qa|grep openssl-develecho "kernel -source"rpm -qa|grep kernel-sourceecho "perl"rpm -qa|grep perlecho "perl -CPAN"rpm -qa|grep perl-CPANecho "cvs"rpm -qa|grep cvsecho "bison"rpm -qa|grep bisonecho "ncurses -devel"rpm -qa|grep ncurses-develecho "audiofile -devel"rpm -qa|grep audiofile-develecho "-"Roman Zhovtulya [EMAIL PROTECTED] wrote: You've got to check if you have all the required mysql libraries installed (mysql client and mysql-devel) -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk asteriskSent: Donnerstag, 3. März 2005 10:13To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk realtime ,asterisk extensions not load form db. Hello ! I try to run asterisk with real time config from database. I use AMP to configure . Everythig it ok , I cansetnew sip and iax extensions, I can see them on mysql db , as well is amp . But these extension I cannt use in asterisk . I have seen some new conf file ;sip_additional.conf;iax_additional.conf;extensions_additional.conf;meetme_additional.conf .If I reload asterisk these extensions are not load . I have these tabele in mysql ,sip, iax, extensions. my extconfig.conf file [settings] ;uncomment to load queues.conf via the db engine.;queues.conf = odbc;conf filename = driver,databasename[,table_name] voicemail.conf = mysql,asterisk,voicemail_table; it is goodsip.conf = mysql,asterisk,sip_tableiax = mysql,asterisk,iaxextensions.conf = mysql,asterisk,extensions How can I set asterisk to load config form database ? Is there anyone who have the same problem ? Thans for help ! Roby. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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] RE: Getting phpconfig to work?
When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; Try a simple php-script in this directory. Something like this, name it test.php ?PHP print (test) ? If it does not print test, and you see the code instead, check your httpd.conf for a general php-parsing problem. True, I have tried that and all I see is code instead. How do I go about solving the php-parsing problem in my httpd.conf then? Rgds, Julius. Regards Guido Hecken ___ 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] RE: Getting phpconfig to work?
True, I have tried that and all I see is code instead. How do I go about solving the php-parsing problem in my httpd.conf then? Try this in your (actual!!) httpd.conf AddType application/x-httpd-php .php Perhaps (not shure) this too, but the above should work # LoadModule php4_module modules/libphp4.so # DirectoryIndex index.php Regards, Guido Hecken ___ 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] Sipura 3000 Inbound Dialing Problem
Thanks Joseph-san I think it my telco problem ?? i checked this with another sipura FXS adaptor connected to FXO port and it works fine with INBAND DTMF configuration. . I will check this with another service provider Thanks a lot Regards Dhananjay SJoseph Finley [EMAIL PROTECTED] wrote: dhananjay sarnaik wrote: Thanks for the information. But still we are facing the same problem. We tried upgrading the firmware to latest available on sipura website and still the result is same. Does any specific DTMF setting required? we have tried all the 3 options in asterisk (inband, rfc2833 and info) but no luck In your SIP.conf make sure it's INBAND and the INBAND is specified on the SIPURA 3000. I had the same problem and that solved it.Joe___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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] Getting phpconfig to work?
DO you have apache2-mod_php installed ? Which distro are you using ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 11:45 AM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Getting phpconfig to work? Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description -- -- Parent Directory03-Mar-2005 12:15 - asterisk.reload 03-Mar-2005 12:28 1k cls_phpconfig.php 03-Mar-2005 11:4814k cls_phpconfig_html.php 03-Mar-2005 11:5517k images/ 24-Feb-2005 09:06 - phpconfig.php 14-Sep-2003 19:32 6k phpconfig_init.php 03-Mar-2005 11:44 2k -- -- Apache/1.3.33 Server at ip-of-machine Port 80 I have made the necessary changes to all the files in the phpconfig directory and my DocumentRoot is set to /usr/local/www/. To add to this, I have php4-4.3.10_1 and apache+mod_ssl-1.3.33+2.8.22 installed on my Asterisk box. What could I be doing wrong? Thanks in advance! Rgds, Julius. ___ 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] Wrong CVS version ?
Hi, I've updated my Asterisk 3 times with : cvs checkout -r v1-0 zaptel asterisk asterisk-addons and then do cd asterisk make clean make make install make samples make progdocs and then when I run Asterisk I get : Asterisk CVS-v1-0-02/11/05-01:46:25, Copyright (C) 1999-2004 Digium. Is this a bug in CVS handling or am I doing something wrong ? How to check which version of CVS I have from commandline ? Regards, Rob. ___ 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] RE: Getting phpconfig to work?
Hi, When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; It's quite likely that your Apache+PHP installation is incomplete / broken. You may want to check that out. Cheers, Gerald. ___ 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] Wrong CVS version ?
you are compiling in wrong sequence first zaptel then asterisk and after that asterisk-addons . hope this helps ___ 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] Wrong CVS version ?
- Original Message - From: Adnan Ahmed [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 1:28 AM Subject: Re: [Asterisk-Users] Wrong CVS version ? you are compiling in wrong sequence first zaptel then asterisk and after that asterisk-addons . hope this helps ??? How this influence on cvs version of asterisk binary ??? Regards, Rob. ___ 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] Re : Calling card platform
We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ 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] Wrong CVS version ?
On Thu, 2005-03-03 at 12:22 +0100, Robert Rozman wrote: Hi, I've updated my Asterisk 3 times with : cvs checkout -r v1-0 zaptel asterisk asterisk-addons and then do cd asterisk rm -f .version make clean make make install -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] how do i get rid of this blasted echo !!!
Any help on this would be great I have 2 TDM400P's, 2 asterisk servers (running on powerful boxes with FC1 and * v CVS 1.0.02), and 4 analogue PSTN lines from BT and whatever I do, I cannot get rid of this damn local echo. Ive tried setting the echoTraining, echoCancel (in phone.conf and Zapata.conf) , echocancelwhenbridged to every possible combination , Ive even tried running the fxotune utility to no avail. Ive swapped cards, telephone lines, servers and also tried different phones (budgetone, x-lite, 7940) but still it's the same. Does anybody have any other suggestions as this is driving me up the wall Cheers Gary ___ 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] Re : Calling card platform
See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ 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] Asterisk realtime , asterisk extensions not load form db.
Yes I've checked . these pakeche I have instaled. But it does not work. echo "libxml2"rpm -qa|grep libxml2echo "libtiff"rpm -qa|grep libtiffecho "libtiff-devel"rpm -qa|grep libtiff-develecho "httpd"rpm -qa|grep httpdecho "mysql"rpm -qa|grep mysqlecho "mysql -devel"rpm -qa|grep mysql-develecho "mysql -server"rpm -qa|grep mysql-serverecho "php"rpm -qa|grep phpecho "php -mysql"rpm -qa|grep php-mysqlecho "openssl"rpm -qa|grep opensslecho "openssl -devel" p;nb sp;rpm -qa|grep openssl-develecho "kernel -source"rpm -qa|grep kernel-sourceecho "perl"rpm -qa|grep perlecho "perl -CPAN"rpm -qa|grep perl-CPANecho "cvs"rpm -qa|grep cvsecho "bison"rpm -qa|grep bisonecho "ncurses -devel"rpm -qa|grep ncurses-develecho "audiofile -devel"rpm -qa|grep audiofile-develecho "-"Roman Zhovtulya [EMAIL PROTECTED] wrote: You've got to check if you have all the required mysql libraries installed (mysql client and mysql-devel) -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk asteriskSent: Donnerstag, 3. März 2005 10:13To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk realtime ,asterisk extensions not load form db. Hello ! I try to run asterisk with real time config from database. I use AMP to configure . Everythig it ok , I cansetnew sip and iax extensions, I can see them on mysql db , as well is amp . But these extension I cannt use in asterisk . I have seen some new conf file ;sip_additional.conf;iax_additional.conf;extensions_additional.conf;meetme_additional.conf .If I reload asterisk these extensions are not load . I have these tabele in mysql ,sip, iax, extensions. my extconfig.conf file [settings] ;uncomment to load queues.conf via the db engine.;queues.conf = odbc;conf filename = driver,databasename[,table_name] voicemail.conf = mysql,asterisk,voicemail_table; it is goodsip.conf = mysql,asterisk,sip_tableiax = mysql,asterisk,iaxextensions.conf = mysql,asterisk,extensions How can I set asterisk to load config form database ? Is there anyone who have the same problem ? Thans for help ! Roby. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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] Re : Calling card platform
Yair, I am honest enough to say what I do , Don't jump into something you don't know...I was working there for a while and that was months ago and it was a part of my freelance contribution. Don't think others to have same kind fruadelent mentality that you have.SO next time before proving yourself very smart from others take the pain to ask what is the matter. Ehsan On Thu, 3 Mar 2005 14:09:06 +0200, Yair Hakak [EMAIL PROTECTED] wrote: See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ 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] wctdm and two tdm cards
* Uncomment if you happen have an early TDM400P Rev H which * sometimes forgets its PCI ID to have wcfxs match essentially all * subvendor ID's I'd love to know how the TJ320 forgets its PCI ID... it's set using resistors, for Chrissakes... Some sort of timing issue in retreiving it? ___ 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: [ gertec ] Re: [Asterisk-Users] timing/clock problem
Yes, I did. It is correct. We have tried with and without crc4. Martijn van Oosterhout wrote: You checked the crc4 setting, right? And the protocols... On Wed, Mar 02, 2005 at 06:49:53PM -0300, Alex G Robertson wrote: But when I configure span4 to get clock source from telco they become unsynchronized. TElco bit rate stays in 2048000 bps, but asterisk stays on 2048443 pbs!! span=4,1,0,ccs,hdb3,crc4 Hope this helps, -- Alex G Robertson NOC - Microlink ___ 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] capi debugging
Hi, Regarding capi debug, I don't know how to translate reasons like 0x3302 or infos like 0.I didn't find any 'translator' googleing capi debugging. Do you know about any 'translator' for this or should I be as clever as to know what a reason 0x3302 is? What is this debug for if I can't interpret it? Kind regards, Victor. From capi debug: == CAPI Call CAPI[contr1/number]/1 -- creating pipe for PLCI=-1 sent CONNECT_REQ MN =0x5 -- CONNECT_CONF ID=002 #0x0005 LEN=0014 Controller/PLCI/NCCI = 0x101 Info = 0x0 -- CONNECT_CONF ID=002 #0x0005 LEN=0014 Controller/PLCI/NCCI = 0x101 Info = 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- Called @number:number -- DISCONNECT_IND ID=002 #0x0009 LEN=0014 Controller/PLCI/NCCI = 0x101 Reason = 0x3302 ___ 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] timing/clock problem
I am using only one sync source. zaptel.conf # ChannelBank 1 span=1,0,0,esf,b8zs fxoks=1-24 # ChannelBank 2 (Empty) span=2,0,0,esf,b8zs fxoks=25-48 # Empty - Loopback is plugged here! span=3,0,0,ccs,hdb3,crc4 bchan=49-63,65-79 dchan=64 # Telco1 - Intelig #span=4,1,0,ccs,hdb3,crc4 - Funcionou com a PRI Telemar sem Alarmes span=4,0,0,ccs,hdb3,crc4 bchan=80-94,96-110 dchan=95 # Ambients fxsks=111-113 And about interruptions it is ok. I'm not sharing it. Andrew Kohlsmith wrote: On March 2, 2005 04:49 pm, Alex G Robertson wrote: When I configure zaptel with no external clock, E1 gets aligned/synchronized with bit rate in 2048000 bps, both me and telco. span=4,0,0,ccs,hdb3,crc4 What are the other spans connected to, and what are their span= lines? I have another PRI circuit used for data connections from Telco2 and I used it for testing. Telco2 (Telemar) do not send alarms and I sync to then. Asterisk cards can only sync to one source per CARD. So if you have a TE410P in there and you're trying to sync to two telcos, it won't work. Now granted, the telcos should all be pretty damn solid for timing and using one over the other shouldn't cause frame slips but this is the nature of the Digium multiport T1/E1 cards. You would need to use two cards (at double the interrupt rate) or try to convince one telco to sync to you. I'd be surprised if you'll get the latter one to occur. :-) -A. ___ 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 -- Alex G Robertson NOC - Microlink ___ 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] how do i get rid of this blasted echo !!!
On March 3, 2005 07:02 am, Brett, Gary wrote: I have 2 TDM400P's, 2 asterisk servers (running on powerful boxes with FC1 and * v CVS 1.0.02), and 4 analogue PSTN lines from BT and whatever I do, I cannot get rid of this damn local echo. Ive tried setting the echoTraining, echoCancel (in phone.conf and Zapata.conf) , echocancelwhenbridged to every possible combination , Ive even tried running the fxotune utility to no avail. Ive swapped cards, telephone lines, servers and also tried different phones (budgetone, x-lite, 7940) but still it's the same. You haven't told us what hardware (platform) you're on, nor have you told us if your FXO ports are in whatever mode they need to be in for UK impedances (I think they default to FCC or North American). For echo on my PRI I could not get rid of it until I recompiled the zaptel and wct4xxp drivers with MMX support enabled and with the instructions reordered and used for the pentium 4 processor (which I'm using, Xeon 2.6 to be exact). After that, the echo magically disappeared. I haven't reverted back to my original (non-processor-optimized, non-MMX-enabled) drivers to see if it comes back, but that's all that's changed and it's in production so I am hesitant to screw around with it any more. -A. ___ 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] Re : Calling card platform
I will make this as clear as i possibly can. 1. i am not very smart from others. I am, however, a big fan of honesty. 2. You WERE NOT honest enough to say what you do. I don't care if you were or are a freelancer, or the CEO, or if they paid you in cows instead of money. You have or had a relationship with the company. You did not mention this. In fact, you painted yourself as a customer, which you might be, but it's not exactly an unbiased recommendation if you used to work there. 3. You can flame me all you want. Evidently i have a fraudulent mentality so it's OK. 4. I'm taking the pain: what is the matter? Were you planning a second email to inform us of your association with the company, did you assume we just all knew you had such an association, did you think it's not relevant? 5. i want to make it clear I know absolutely nothing about this product. It might be really great. Please keep in mind that my anger is at the poster and not the company. -yair On Thu, 3 Mar 2005 18:21:40 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: Yair, I am honest enough to say what I do , Don't jump into something you don't know...I was working there for a while and that was months ago and it was a part of my freelance contribution. Don't think others to have same kind fruadelent mentality that you have.SO next time before proving yourself very smart from others take the pain to ask what is the matter. Ehsan On Thu, 3 Mar 2005 14:09:06 +0200, Yair Hakak [EMAIL PROTECTED] wrote: See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ 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] Wrong CVS version ?
On Thu, 2005-03-03 at 13:23 +0100, Robert Rozman wrote: On Thu, 2005-03-03 at 12:22 +0100, Robert Rozman wrote: Hi, I've updated my Asterisk 3 times with : cvs checkout -r v1-0 zaptel asterisk asterisk-addons and then do cd asterisk rm -f .version Thanks for info. But, is this normal to do it by hand ? Well, I have a little script that does it for me. Or, you could use: cd asterisk make update Which I think will do the cvs update and rm .version for you. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Installing modules for TDM400p
I have linux 2.6.5 running on my machine.I downloaded The latest version of Zaptel from the cvs repoistory.Compiled zaptel with the make linux26 option. Installed it by modprobe which gave no errors.However when i did modprobe wctdm i got the following error. FATAL: Module wctdm not found. I have no idea why it's happening.Tried Googling but got nothing.ANY Suggestions?? ___ 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] Getting phpconfig to work?
No, I have apache 1.3.33 and mod_ssl 2.8.22 installed. Do I need to have apache2-mod_php installed? Rgds, Julius. DO you have apache2-mod_php installed ? Which distro are you using ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 11:45 AM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Getting phpconfig to work? Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description -- -- Parent Directory03-Mar-2005 12:15 - asterisk.reload 03-Mar-2005 12:28 1k cls_phpconfig.php 03-Mar-2005 11:4814k cls_phpconfig_html.php 03-Mar-2005 11:5517k images/ 24-Feb-2005 09:06 - phpconfig.php 14-Sep-2003 19:32 6k phpconfig_init.php 03-Mar-2005 11:44 2k -- -- Apache/1.3.33 Server at ip-of-machine Port 80 I have made the necessary changes to all the files in the phpconfig directory and my DocumentRoot is set to /usr/local/www/. To add to this, I have php4-4.3.10_1 and apache+mod_ssl-1.3.33+2.8.22 installed on my Asterisk box. What could I be doing wrong? Thanks in advance! Rgds, Julius. ___ 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] More NAT questions -- SOLVED
Hi, all Got it to work finally. Thanks to all. Had to add [general] externip=xxx.xxx.xxx.xxx ;ip address of your nat firewall (public ip) localnet=192.168.0.0/24; the local subnet where the asterisk box is Actually, I had 'externip' before, but I have added 'localnet' one. I also had to do port forwarding on the NAT near to PHONE 2 to pass port 5060 to the phone. This is needed if you ever want to call this phone. I can e-mail my sip.conf to anyone who is interested. Rudolf - Original Message - From: Julian J. M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 03, 2005 4:11 AM Subject: Re: [Asterisk-Users] More NAT questions In you asterisk sip.conf: [general] externip=xxx.xxx.xxx.xxx ;ip address of your nat firewall (public ip) localnet=192.168.0.0/24; the local subnet where the asterisk box is If you don't externip, externip will never be used, because asterisk won't know WHEN to use it. Also, define canreinvite=no in your sip phones sections, as was suggested above. Julian J. M. On Wed, 2 Mar 2005 23:26:56 +1100, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Still trying to get NAT working. I have following setup: PHONE 1 -- * BOX | NAT/Firewall | | NAT/Firewall | | PHONE 2 Firewall next to phone 2 has all ports open. Firewall next to Asterisk has open ports 5060 and 1:2. All of those are forwarded to Asterisk box. Both phones succesfully register with Asterisk. (I had to add NAT=yes to configuration of PHONE 2 in sip.conf to get this far). Now, problems: I can place a call from PHONE2 to PHONE1, but sound path is not established. Calls from PHONE1 to PHONE2 can not be placed at all. (I assume that this is because port 5060 is not forwarded to the phone at NAT/Firewall, but more on it later). Looking at SIP debug info, Asterisk tries to use local address of PHONE2 instead of its public IP. As a result, no info can be sent to it. I have tried to install SIPROXD on the NAT/Firewall close to Asterisk box, but this did not help. Now, we have tried to use one of the commercial VoIP service at PHONE2 location. We had to use their phone and it worked just fine without any alterations to NAT/Firewall device. I am pretty sure that they use SIP, so they did resolve the problem somehow. Sorry, there is no technical info available on this service. Did anyone succeeded in doing this setup? I know, IAX is a better way, but I can not setup many Asterisk boxes. Basically, I am doing it for a friend. He is working for a small medical company. They have number of offices that are not open every day and offices are too small to put Asterisk box in each one. There will be 1-3 IP phones in each office, except central one. Central one will need Asterisk, the rest should be on their own. Any help is greatly appreciated. Thanks, Rudolf ___ 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] Re: Installing modules for TDM400p
Also when i had compiled zaptel i had got the following message: *** Warning: zt_register [/usr/src/zaptel/wctdm.ko] has no CRC! *** Warning: zt_qevent_lock [/usr/src/zaptel/wctdm.ko] has no CRC! *** Warning: zt_hooksig [/usr/src/zaptel/wctdm.ko] has no CRC! *** Warning: zt_ec_chunk [/usr/src/zaptel/wctdm.ko] has no CRC! *** Warning: zt_transmit [/usr/src/zaptel/wctdm.ko] has no CRC! *** Warning: zt_receive [/usr/src/zaptel/wctdm.ko] has no CRC! *** Warning: zt_unregister [/usr/src/zaptel/wctdm.ko] has no CRC! On Thu, 3 Mar 2005 17:43:40 +0500, Rizwan Chaudhry [EMAIL PROTECTED] wrote: I have linux 2.6.5 running on my machine.I downloaded The latest version of Zaptel from the cvs repoistory.Compiled zaptel with the make linux26 option. Installed it by modprobe which gave no errors.However when i did modprobe wctdm i got the following error. FATAL: Module wctdm not found. I have no idea why it's happening.Tried Googling but got nothing.ANY Suggestions?? ___ 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] Getting phpconfig to work?
No, I have apache 1.3.33 and mod_ssl 2.8.22 installed. Do I need to have apache2-mod_php installed? If you see the source of the script, you don't have php installed or configured correctly. go read this : http://www.php.net/manual/en/install.unix.php hth ___ 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] Getting phpconfig to work?
There should be and apache_mod-php package if using RH related ditro. apache2-mod_php is for Apache 2 and above if I'm not mistaken. Which ditribution of Linux are you using. Red Hat, Mandrake, Debian, Gentoo ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 02:51 PM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Getting phpconfig to work? No, I have apache 1.3.33 and mod_ssl 2.8.22 installed. Do I need to have apache2-mod_php installed? Rgds, Julius. DO you have apache2-mod_php installed ? Which distro are you using ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 11:45 AM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Getting phpconfig to work? Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description -- -- Parent Directory03-Mar-2005 12:15 - asterisk.reload 03-Mar-2005 12:28 1k cls_phpconfig.php 03-Mar-2005 11:4814k cls_phpconfig_html.php 03-Mar-2005 11:5517k images/ 24-Feb-2005 09:06 - phpconfig.php 14-Sep-2003 19:32 6k phpconfig_init.php 03-Mar-2005 11:44 2k -- -- Apache/1.3.33 Server at ip-of-machine Port 80 I have made the necessary changes to all the files in the phpconfig directory and my DocumentRoot is set to /usr/local/www/. To add to this, I have php4-4.3.10_1 and apache+mod_ssl-1.3.33+2.8.22 installed on my Asterisk box. What could I be doing wrong? Thanks in advance! Rgds, Julius. ___ 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 Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: No, I have apache 1.3.33 and mod_ssl 2.8.22 installed. Do I need to have apache2-mod_php installed? Rgds, Julius. DO you have apache2-mod_php installed ? Which distro are you using ? Content analysis details: (0.1 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO 0.0 LOTS_OF_STUFF BODY: Thousands or millions of pictures, movies, etc. ___ 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] RE: Getting phpconfig to work?
I have done so and it returns the very code I created that is; ?php phpinfo(); ? Secondly, which php are you referring to? Is it php4 or mod_php? And if so, which of the two do I need for this? Are you sure you have php correctly installed. Do the test page as below and let us know what it comes up with -Original Message- From: World Wide Web Owner [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 10:17 To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: RE: Getting phpconfig to work? Hi, Does phpconfig require a particular php package installed? I have php4-4.3.10_1 installed on my box. Does this have an effect? What do I need to change in terms of ownership and permissions to files located in the phpconfig directory? At the moment I have, drwxr-xr-x 4 root wheel512 Mar 3 13:09 . drwxr-xr-x 8 root wheel512 Mar 3 12:15 .. drwxr-xr-x 2 root wheel512 Feb 24 09:06 CVS -rwxrwxrwx 1 root wheel588 Mar 3 12:28 asterisk.reload -rw-r--r-- 1 root wheel 14702 Mar 3 11:48 cls_phpconfig.php -rw-r--r-- 1 root wheel 17846 Mar 3 11:55 cls_phpconfig_html.php drwxr-xr-x 3 root wheel512 Feb 24 09:06 images -rw-r--r-- 1 root wheel 6515 Sep 14 2003 phpconfig.php -rw-r--r-- 1 root wheel 2128 Mar 3 11:44 phpconfig_init.php Any indicators? Rgds, Julius. That means PHP isn't setup properly on your server; it isn't parsing the file as PHP. It is just outputting the code. Create a file named test.php and put this in the file: ?php phpinfo(); ? This should display lots of information in a table about your php installation. When you have this working; phpconfig should work. C -Original Message- From: World Wide Web Owner [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 10:02 To: C. Tomlinson Subject: RE: Getting phpconfig to work? Hi, When I do click on the phpconfig.php link, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output from http://ip-of-machine/phpconfig/phpconfig.php below; ?PHP /** * * Asterisk configuration file interface script * * * * * * phpconfig:,v 1.0 2003/07/03 17:19:37 * Authors: Dave Packham [EMAIL PROTECTED] * Rob Birkinshaw [EMAIL PROTECTED] */ require_once(phpconfig_init.php); require_once(cls_phpconfig.php); require_once(cls_phpconfig_html.php); // create and initalize objects // $page = new Open_Conf_HTML(); $conf = new Open_Conf(); // setup 3pane html theme $page-OC_HTML_set3Pane(true); // pass thru phpconfig_init paramaters to object $conf-OC_setConfDirectories($conf_directories); $conf-OC_setTempDir($temporary_directory); $conf-OC_setTempPrefix($temporary_file_prefix); $conf-OC_setAccessFile($access_file); $conf-OC_setConfDir($default_conf_file_direcotry,$conf_directories); $conf-OC_setConfFilter($conf_file_filter); $conf-OC_setResetCmd($reset_cmd); $conf-OC_setRemark($remark); // pass thru html output parameters $page-OC_HTML_setTextareaRows($textarea_rows); $page-OC_HTML_setImagesDir($images_dir); $page-OC_HTML_setLogo($logo); $page-OC_HTML_setTitle($title); $page-OC_HTML_setPageTitle($page_title); $page-OC_HTML_setDescription($description); $page-OC_HTML_setKeywords($keywords); $page-OC_HTML_setWebmaster($webmaster); $page-OC_HTML_setDisclaimer($disclaimer); $page-OC_HTML_setFooterText($footer_text); $page-OC_HTML_setLogoLink($logo_link); // init top menu bar $page-OC_HTML_setHeaderBar(array_merge($conf-_OC_conf_dirs , array(Re-Read Configs=phpconfig.php?reset=reset))); // init side menus $menuList = array(); $rightMenuList = array(); session_start(); // this session variable will be set by a login screen // in a future release // for now, fake it for the prototype $_SESSION['valid_user'] = $fakeuser; $conf-OC_checkValidUser(); if($_GET['file']) // conf file requested via menu How do I proceed from here? Thanks, Julius. Hi, You need to click on the phpconfig.php to actually load the page. It is because the file is not called index.php. This could be changed in the apache conf file if wished. Let us know how you go. C -Original Message- From: World Wide Web Owner [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 03 March 2005 09:45 To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: Getting phpconfig to work? Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description
Re: [Asterisk-Users] capi debugging
I'm going to answer myself. I don't know If somebody already did it because I'm using digest mode. CAPI specification is available at http://www.capi.org/, It explains all the commands and associated identifiers. Now I know that reason0x3302 in DISCONNECT_IND means Protocol error, Layer 2. I'llcarry on with myresearching from here.I don't know what is the point of use messages like 0x3302 instead of speak a human languagebut I've found my 'translator'. Victor. ___ 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] Way to disable # as transfer and just take thekey.
This is quite possibly the most popular question on this mailing list. - Remove t/T options from the Dial command this only works in CVS-HEAD not 1.0x stable - Change the transfer-key in features.conf ___ 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] Calling hangup in background
Hi everybody, I'm running an IVR menu with different languages setted up by user when they enter this menu. What I want is when they hangup, asterisk sets the default language (aka en) back. I'm wondering which extension is called after a hangup in a background command? BTW my IVR menu is in a goto context. -- Daniel ___ 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] (another try) Dialing phone number and extension together to avoid listening to voice menu (incoming call)
Hello,
Sorry for reposting the message, but I'm not sure the first post went
through.
I'm trying to figure out how to get Asterisk to dial an extension when a
call comes from the outside and contains the extension already.
(Somebody wants to call a user of Asterisk with extension 111 from the
outside)
For example: I've hooked Asterisk to sipgate.de and received a landline
phone number (say 0781205237).
Now if you dial 0781205237 and and an extension altogether
(0781205237111) I would like Asterisk to redirect the call to the
extension 111, without having to listen to the greetings message and
then typing the extension on the keypad.
Please help me to figure it out. Any suggestions and code excerpts would
be highly appreciated.
Also, I was trying to use a voice menu setup for that, so that when the
user dials 0781205237, he/she would listen to the greeting and then
can enter the extension on the phone. However, I couldn't get this to
work either.
Here is the excerpt of my extensions.conf:
***
; defining the voice menu for incoming calls:
[fhostaffmenu]
exten = s,1,Ringing; Make them comfortable with
some seconds of ringback
exten = s,2,Answer ; Answer the line
exten = s,3,DigitTimeout(1); Set Digit Timeout to 5 seconds
exten = s,4,ResponseTimeout(3) ; Set Response Timeout to 10 seconds
exten = s,5,Read(mynumber,beep,3) ; Read DTMF input and save it into
mynumber variable exten =
s,6,Macro(fhostaff,${mynumber},SIP/${mynumber}) ; dial the extension
that is saved in mynumber
***
When I execute this, it says that User entered ''. Why wouldn't it
read the numbers punched on the phone? The Voicemail works very well.
I use dtmfmode = rfc2833 and iLBC codec.
Also, please check if the comments I made to the code below are correct.
Thank you very much,
Roman Zhovtulya
___
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] Calling hangup in background
Try adding an exten = h,1,DoSomething in the context Jason On Thu, 03 Mar 2005 14:40:10 +0100, administrator tootai [EMAIL PROTECTED] wrote: Hi everybody, I'm running an IVR menu with different languages setted up by user when they enter this menu. What I want is when they hangup, asterisk sets the default language (aka en) back. I'm wondering which extension is called after a hangup in a background command? BTW my IVR menu is in a goto context. -- Daniel ___ 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] ASTCC questions
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ronald
Wiplinger
Sent: Thursday, March 03, 2005 2:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ASTCC questions
Ronald Wiplinger wrote:
(Correcting my own message)
I have setup ASTCC as:
trunk:
NuFone IAX2 NuFone
should be:
NuFone IAX2 [EMAIL PROTECTED] !!!
1. So far I can call out, but I cannot call in. - Any hints?
2. ASTCC shows me for my test calls only:
In Cards that I used from 1 60 pennies
If I try to get detail info from the card, than I get:
/Card /*886228803959 */ has used /*60* of *10* units
Caller*ID Called Number Trunk Disposition Billable Seconds
Billed Cost
but no detail data!!!Any hints???
Ronald,
The CVS ASTCC has an error in the database table structure for the call
records.
See http://bugs.digium.com/bug_view_page.php?bug_id=0002796
for a patch to the cgi scripts that create the table. Basically, the
callstart field is missing in the
CREATE table cdrs statement.
The above link also has a few additions to ASTCC that may be interesting to
you. Specifically,
there is an extension that allows you to use the caller id as the account
number but also require a
PIN to complete the call.
Karl Putz
bye
Ronald
routes:
^1415.* California NuFone 0 0 200
iax.conf
=
register = User:[EMAIL PROTECTED]
[NuFone]
type=peer
host=switch-1.nufone.net
secret=my_secret
[NuFone]
type=user
secret=my_secret
context=fromNuFone
extensions.conf
==
[NuFone]
exten =
_91NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
exten = _9011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
With above settings I see in CLI when I am dialing:
-- Executing NoOp(SIP/886228803959-1e6d, SetCallerID()) in new
stack
-- Executing Dial(SIP/886228803959-1e6d,
IAX2/[EMAIL PROTECTED]/14159625000) in new stack
-- Called [EMAIL PROTECTED]/14159625000
-- Call accepted by 66.225.202.72 (format ulaw)
-- Format for call is ulaw
-- IAX2/NuFone-11 answered SIP/886228803959-1e6d
-- Hungup 'IAX2/NuFone-11'
== Spawn extension (VoIP_customer_Phone, 914159625000, 2) exited
non-zero on 'SIP/886228803959-1e6d'
It works !!!
Changing the settings in extensions.conf to:
[NuFone]
;exten =
_91NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
;exten = _9011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
;
exten = _91NXXNXX,1,NoOp(SetCallerID(${username}))
exten =
_91NXXNXX,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,3,hangup
;
exten = _9011N.,1,NoOp(SetCallerID(${username}))
exten = _9011N.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:${TRUNKMSD}})
exten = _9011N.,3,hangup
gives me in CLI by redialing the same number:
-- Executing NoOp(SIP/886228803959-e043, SetCallerID()) in new
stack
-- Executing DeadAGI(SIP/886228803959-e043,
astcc.agi|886228803959|14159625000) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
-- Playing 'digits/10' (language 'en')
-- Registered IAX2 to '69.73.19.178', who sees us as
61.220.121.20:4569
-- Playing 'digits/2' (language 'en')
-- AGI Script Executing Application: (DIAL) Options:
(IAX2/NuFone/14159625000|30|HL(1998:6:3))
-- Limit Data:
-- timelimit=1998
-- play_warning=6
-- play_to_caller=yes
-- play_to_callee=no
-- warning_freq=3
-- start_sound=UNDEF
-- warning_sound=timeleft
-- end_sound=UNDEF
-- Called NuFone/14159625000
Mar 3 14:00:31 WARNING[8102]: chan_iax2.c:6280 socket_read: Call
rejected by 66.225.202.72: No such context/extension
-- Hungup 'IAX2/NuFone-3'
== No one is available to answer at this time (1:0/0/0)
-- AGI Script astcc.agi completed, returning 0
-- Executing Hangup(SIP/886228803959-e043, ) in new stack
== Spawn extension (VoIP_customer_Phone, 914159625000, 3) exited
non-zero on 'SIP/886228803959-e043'
Why it tells me: No such context/extension ???
What do I need to change?
Thanks for your help in advance!
bye
Ronald
___
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
--
Ronald Wiplinger (CEO of ELMIT)
http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208
- I'm a SpamCon Foundation Member, #694, Verify it at
http://www.spamcon.org
PS: Spam prevention!
Our system is
protected with a spam prevention program.
If you send us an e-mail, our system will send you a confirmation
message back. Just reply to this confirmation message please.
After receiving this confirmation message, our system will send
the hold message (one) and all future
Re: [Asterisk-Users] What my IAXy could have been...
Quoted Message 11:10 05/03/01 -0800 from Ed Greenberg: Sipura 1000 or 2000? Thanks, Ed. Just looked at the Sipura 1000 specs again - but there's no mention of the IAX protocol anywhere. Now, i hadn't mentioned that in my original message (somehow i assumed this list was about IAX related hard and software), so i am certainly not complaining about your answer - the Sipura 1000 specs appear to be pretty competent. :-) But i AM looking for info on another IAX capable device - like the IAXy, but more user friendly, as it were... Thanks regards: H. D. --On Tuesday, March 01, 2005 10:15 PM +0900 Daiku [EMAIL PROTECTED] wrote: [...] What would you recommend as a rugged, small, and easy to configure/use self-contained unit that one could carry along in one's hand luggage when traveling and plug into the LAN wherever someone has an ADSL, cable, or fibre-optic connection? A VoIP unit for all seasons? [...] ___ 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] Some errors on sip debug
I have some problem to configure the call from asterisk to ser.[globals]SERADDRESS=xxx.xxx.xxx.xxx:5060exten = 77,1,Dial(SIP/[EMAIL PROTECTED],20,r)Error in Sip Debug ---NOTICE[25541]: chan_sip.c:6848 handle_response: Failed to authenticate on INVITE to '"Alexg" sip:[EMAIL PROTECTED]:5061;tag=as3cf27769'Any help will be appreciated.Thanks__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Update on the blending of app_cbmysql and app_meetme2 (out of tree modules)
Once again, I know that a true MeetMe2 is in the works, but if anyone is looking for a database conferencing setup then check this out. New diffs and PHP web interfaces at www.fitawi.com/Asterisk -New in the interface: Add a conference Modify an existing conference -New in app_cbmysql: Renamed confmysql.conf to cbmysql.conf Move conference application name to conf file Move conference count application name to conf file Moved compile time conference options into the conf file -ToDo in the interface: Add a delete function to web interface Add conference ID/time conflict detection to web interface Migrate date/time entry to listbox or calendar pop-up -ToDo in app_cbmysql: The code is munging the channel name into the conference ID This doesn't impact functionality, but is ugly and quite possibly something I screwed up Use Caller-ID for user_id An interface to mute or boot callers kind of needs this -ToDo in app_meetme2 Port recent MeetMe bugfixes and enhancements Add conference termination code Add conference ending warning Original source: App_MeetMe2 http://www.areski.net/asterisk-meetme/about.php?s=0 App_CBMysql http://www.mithotech.com/asterisk/ Comments, tips on PHP or suggestions welcome. Dan ___ 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] Cannot reset an IAXy box!!!
Quoted message 13:01 05/02/13 -0600, from Eric Wieling: The IAXy does not use DHCP, it uses the older BOOTP protocol. Most DHCP servers support BOOTP (but it may have to be enabled) Eric, that's gotta be one of the most useful bits of information about the IAXy i've seen in a long while... ;-) Where on the related websites did i miss that? -- ___ 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] CDR
I need that my records cdr only get the calls that begin with 9 or any other rule is this possible?? thanks in advance wert Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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] Getting phpconfig to work?
Julius Kidubuka wrote: No, I have apache 1.3.33 and mod_ssl 2.8.22 installed. Do I need to have apache2-mod_php installed? It does not necessarily have to be apache2-mod_php but you definitely need to have mod_php installed. Cheers, Eric ___ 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] IAXy and Private IP
Hi everyone, I'm going to install an IAXy in my client's office but there the internet conection has a private IP address, however the Asterisk server has a public IP. Private IP Public IP (IAXy)(ISP LAN)---((Internet))-(Asterisk Server) This setup can cause any problems to the comunication process? I'm aware that the IAX2 protocol is NAT friendly so I think this will work, but to be sure I want to hear some oppinions. Thanks for all, Raul. ___ 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] (another try) Dialing phone number and extension together to avoid listening to voice menu (incoming call)
***
; defining the voice menu for incoming calls:
[fhostaffmenu]
exten = s,1,Ringing ; Make them comfortable with
some seconds of ringback
exten = s,2,Answer ; Answer the line
You haven't actually given them any ringing, you need to add this:
exten = s,3,wait(2) ; Give them 2 seconds of ringing
exten = s,4,DigitTimeout(1) ; Set Digit Timeout to 5 seconds
exten = s,5,ResponseTimeout(3) ; Set Response Timeout to 10 seconds
Rather than doing the below, if you simply stop all processing at this
point, and don't have any more extensions, then asterisk will wait 3
seconds for the user to press a number, then 1 second for each extra
number. When they don't press a number for more than the 1 second, or
asterisk matches an extension, then it will try to dial the entered
number.
exten = s,5,Read(mynumber,beep,3) ; Read DTMF input and save it into
mynumber variable exten =
s,6,Macro(fhostaff,${mynumber},SIP/${mynumber}) ; dial the extension
that is saved in mynumber
***
OK, hard to get asterisk to do this, but something like:
exten = _XXX.,Macro(fhostaff,${mynumber},SIP/${mynumber})
So, the user can dial 3 or more digits, and then it will go to your
macro.
You can also add:
exten =
i,1,playback(some_file_to_say_you_entered_an_invalid_extension_try_again)
exten = i,2,Goto(s,4)
and also:
exten =
t,1,playback(some_file_to_say_you_did_not_enter_an_extension_try_again)
exten = t,2,Goto(s,4)
I hope that helps you...
Regards,
Adam
--
--
Adam Goryachev
Website Managers
Ph: +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au
___
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] Calling hangup in background
Jason Williams a écrit : Try adding an exten = h,1,DoSomething in the context I was looking on mars what I had on my noze! Thanks, it did it. Jason On Thu, 03 Mar 2005 14:40:10 +0100, administrator tootai [EMAIL PROTECTED] wrote: Hi everybody, I'm running an IVR menu with different languages setted up by user when they enter this menu. What I want is when they hangup, asterisk sets the default language (aka en) back. I'm wondering which extension is called after a hangup in a background command? BTW my IVR menu is in a goto context. ___ 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] CDR
Am Donnerstag 03 März 2005 14:59 schrieb R A: I need that my records cdr only get the calls that begin with 9 or any other rule is this possible?? yes Jens ___ 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] CDR
hi, you need to tell us how you're saving your cdr's - database, csv, whatever?- if you're saving to a database a stored procedure is probably best, unless you want to change the SQL statements in the proper module. yair On Thu, 3 Mar 2005 05:59:59 -0800 (PST), R A [EMAIL PROTECTED] wrote: I need that my records cdr only get the calls that begin with 9 or any other rule is this possible?? thanks in advance wert Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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] What my IAXy could have been...
you and everyone else :-) From: Daiku [mailto:[EMAIL PROTECTED] But i AM looking for info on another IAX capable device - like the IAXy, but more user friendly, as it were... ___ 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] problem registering a bt100 with 1.0.5.11 firmware
hi all I can not register my new granstream bt100 phone with asterisk, i have old of they working perfectly but they have an older firmware(1.0.5.3). anybady now where i can read about this or now what i have to do??? My sip.conf: [10] type=friendcontext=unr username=10callerid=10usecallerid=yeshidecallerid=nocanreinvite=yeshost=dynamic dtmfmode=infonat=nomailbox=10callgroup=1pickupgroup=1threewaycalling=yestransfer=yesinmediate=nocancallfoward=yesdisallow=allallow=gsmallow=ulaw thanks in advance wert Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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] [OT] stupid firmware question...
tim panton wrote: Snom do. At least there is this link on their website I must tip my hat to Snom for that - had I noticed it before advising our purchase of the Cisco's we probably would have Snom 220's right now. Oh well, back to seeing just how high I can make my Cisco phone jump. -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
Re: [Asterisk-Users] wctdm and two tdm cards
Matt Gibson wrote:
Still not working -
I did notice something kinda weird tho, After adding
{ 0xe159, 0x0001, 0xa900, PCI_ANY_ID, 0, 0, (unsigned long) wctdmh },
to wctdm.c, and rebooting
when I issue lspci -v, the PCI id on the card has changed (?). Is this
a normal thing to happen?
Instead of being 0xa900 it's now seeing as a9fd:0003
I havent changed anything cept rebuilt the zaptel source.
Matt
ps: none of the methods mentioned have worked so far.
You might want to send this off to Digium
___
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] wctdm and two tdm cards
Matt Gibson wrote: I tried this, but I think this message is slightly outdated, as In my wctdm.c (not wcfxs.c) I have the following, which leads me to believe that it should be already incorporated. Yeah the file name has changed, but the concept is still valid. Is there some way to send a command to the card on reboot to signify that it has lost power and should come back up or something? I'm not familiar at all with low level driver programming or anything, but just a thought. Not that I am aware regards ___ 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] country/city codes
Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 Does anyone know a formula for determining which part ofa dialled number is thecountry code and city code ? ___ 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] registering ALL ser users
Hi I have setup sip.conf and extensions to accept and route calls to voicemail from SER, but I have a strange problem, if the user in xlite has username iqbal, then asterisk needs to have such a user in sip.conf, which in turn would mean that all my users would need in there. All my users are registered at ser, how can I get asterisk to accept all inbound as long as they are from my ser proxy tks Iqbal ___ 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] Zyxel Prestige 2000W
Yes. It works fine here. Be sure to upgrade to the newest firmware. Our phones came with (what seemed like) the oldest. Works fine with our 64bit WEP. -Matthew - Original Message - From: Rod Bacon [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, March 02, 2005 8:28 PM Subject: [Asterisk-Users] Zyxel Prestige 2000W Does anyone have this piece of crap working properly with Asterisk? I thought my problems were all due to NAT, but even on my local LAN segment it's still flaky. Symptoms include periodically losing registration and/or being able to make/receive one call, then not another until rebooted (failing to hang-up). P.S. It's NOT a wireless problem. I'm sitting right on top of the access point, and have a strong, clean signal. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ 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] country/city codes
To my knowledge, there is no such formula. However, you can obtain a database of the entire ITU E164 numbering plan at http://www.numberingplans.com, which have an updated database of all that information. Nir S From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Services Sent: Thursday, March 03, 2005 4:25 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] country/city codes Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 Does anyone know a formula for determining which part ofa dialled number is thecountry code and city code ? ___ 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] Re : Calling card platform
I am a customer and I am paying them every month. I was giving out my personal opinions about the soft . What's so wrong with that if I had not said that I worked with this company 5 months ago ? Don't you have your eyes and judgements before you can buy the product ? So as you know I wokred with them it makes me a fraud or changes the whole software? Please tell me the impact of knowing I worked with them 5 months ago ? I think what you have done so far is not decent enough . You have the right the say anything but which are fact and you know it to be. It is something if I say you have your own platform and you are jealous to let know others about a good platform. I think all the people here are matured enough to get their judgements on the product rather than jsut ordering it because I said so. Ehsanul Karim On Thu, 3 Mar 2005 14:31:56 +0200, Yair Hakak [EMAIL PROTECTED] wrote: I will make this as clear as i possibly can. 1. i am not very smart from others. I am, however, a big fan of honesty. 2. You WERE NOT honest enough to say what you do. I don't care if you were or are a freelancer, or the CEO, or if they paid you in cows instead of money. You have or had a relationship with the company. You did not mention this. In fact, you painted yourself as a customer, which you might be, but it's not exactly an unbiased recommendation if you used to work there. 3. You can flame me all you want. Evidently i have a fraudulent mentality so it's OK. 4. I'm taking the pain: what is the matter? Were you planning a second email to inform us of your association with the company, did you assume we just all knew you had such an association, did you think it's not relevant? 5. i want to make it clear I know absolutely nothing about this product. It might be really great. Please keep in mind that my anger is at the poster and not the company. -yair On Thu, 3 Mar 2005 18:21:40 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: Yair, I am honest enough to say what I do , Don't jump into something you don't know...I was working there for a while and that was months ago and it was a part of my freelance contribution. Don't think others to have same kind fruadelent mentality that you have.SO next time before proving yourself very smart from others take the pain to ask what is the matter. Ehsan On Thu, 3 Mar 2005 14:09:06 +0200, Yair Hakak [EMAIL PROTECTED] wrote: See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ 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] iax notransfer=no and Tt in Dial()
I've not heard anything about this from anyone. I'm taking that to mean that I'm unique in having this problem. I think I will upgrade to a newer version of * and try again. I will report back with more questions or the solution. Thanks. On Tue, Mar 01, 2005 at 09:18:00PM -0500, Michael George wrote: I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set notransfer=no in the iax.conf file at the office so that the office system can step out of the media path and save a hop. That also works fine. However, that does not allow me to transfer someone who called my home extension at the office to someone else at the office. I have put the T/t options in the dial() command as I should. However, the office * box will still transfer the call, stepping out of the media path and breaking my ability to do the intra-office transfer. According to what I find in teh mailing list archives, putting a T/t as an option to dial() will halt a possible transfer and keep the system in the media path. However, that doesn't seem to be the case. I ran asterisk -vvvr to watch the call being processed and I can see the DIAL(channel||T) be called and shortly thereafter it gives the Ready to transfer and then indicates the hangup while the other two * systems are handling the channel. So what I see happening is not what the docs and archives say should be happening. Is this a new feature, that notransfer=no trumps T/t in the dial() command? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] Re: Dial application invoked again and again
Kamran Ahmad wrote: hi If i remove _. from my dialplan(extensions.conf). application is invoked only once. otherwise application is invoked again and again. any one know what is the problem and how to make (global) dialplan for all user agents. When a call hangs up Asterisk will loog for an 'h' extension. _. will match h and so will run again and again. Don't be lazy, dialplans are hard work and you can't just use _. as a pattern and expect it to work. What is the shortest number you might call? If it's 7 digits than use _XX. as your pattern. if it's only 2 digits, use _X. Since X means any number that pattern will not match extension h. ___ 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] CDR
Hi, I think you are going the wrong way, let asterisk register all the calls, and then simply query accordingly. In example, lets say you use the MySQL CDR backend, after all the CDR's are in the DB, simply run: 'SELECT * from cdr where dialednumber like 9% order by calldate asc' That should give you the CDR's you wish for. Of course that is based on the fact that you are logging the CDR's to a database. If you log to a CSV file, you can make a PERL parser to analyze it and bring out the proper CDR's. JMHO on the subject. Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Thursday, March 03, 2005 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CDR hi, you need to tell us how you're saving your cdr's - database, csv, whatever?- if you're saving to a database a stored procedure is probably best, unless you want to change the SQL statements in the proper module. yair On Thu, 3 Mar 2005 05:59:59 -0800 (PST), R A [EMAIL PROTECTED] wrote: I need that my records cdr only get the calls that begin with 9 or any other rule is this possible?? thanks in advance wert Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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
Re: [Asterisk-Users] Getting phpconfig to work?
Hi, I have managed to re-install apache and php. I tried to install mod_php but it failed and returned the error below; === mod_php4-4.3.10_1,1 conflicts with installed package(s): php4-4.3.10_1 They install files into the same place. Please remove them first with pkg_delete(1). *** Error code 1 Stop in /usr/ports/www/mod_php4. *** Error code 1 Stop in /usr/ports/www/mod_php4. I decided to leave it out and when on to setup all the files under the phpconfig directory. When I tried to access http://ip-of-machine/phpconfig/, I got the following error; Fatal error: Call to undefined function: session_start() in /usr/local/www/phpconfig/phpconfig.php on line 57 Could this error be due to not having installed mod_php? If not, then how do I proceed? Thanks in advance! -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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] Re : Calling card platform
It is fine to tout your own products. we call that marketing. However, anyone who claims that they can endorse a product and not mention that they worked for the manufacturer 5 months ago, and thinks this is an ethical thing to do, is not worth my time. Once again, i don't care about the platform. It's probably a very good platform. Your recommendation does not change anything about the platform, but it does call your integrity as a recommender into question. This conversation is over. -yair On Thu, 3 Mar 2005 20:41:55 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: I am a customer and I am paying them every month. I was giving out my personal opinions about the soft . What's so wrong with that if I had not said that I worked with this company 5 months ago ? Don't you have your eyes and judgements before you can buy the product ? So as you know I wokred with them it makes me a fraud or changes the whole software? Please tell me the impact of knowing I worked with them 5 months ago ? I think what you have done so far is not decent enough . You have the right the say anything but which are fact and you know it to be. It is something if I say you have your own platform and you are jealous to let know others about a good platform. I think all the people here are matured enough to get their judgements on the product rather than jsut ordering it because I said so. Ehsanul Karim On Thu, 3 Mar 2005 14:31:56 +0200, Yair Hakak [EMAIL PROTECTED] wrote: I will make this as clear as i possibly can. 1. i am not very smart from others. I am, however, a big fan of honesty. 2. You WERE NOT honest enough to say what you do. I don't care if you were or are a freelancer, or the CEO, or if they paid you in cows instead of money. You have or had a relationship with the company. You did not mention this. In fact, you painted yourself as a customer, which you might be, but it's not exactly an unbiased recommendation if you used to work there. 3. You can flame me all you want. Evidently i have a fraudulent mentality so it's OK. 4. I'm taking the pain: what is the matter? Were you planning a second email to inform us of your association with the company, did you assume we just all knew you had such an association, did you think it's not relevant? 5. i want to make it clear I know absolutely nothing about this product. It might be really great. Please keep in mind that my anger is at the poster and not the company. -yair On Thu, 3 Mar 2005 18:21:40 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: Yair, I am honest enough to say what I do , Don't jump into something you don't know...I was working there for a while and that was months ago and it was a part of my freelance contribution. Don't think others to have same kind fruadelent mentality that you have.SO next time before proving yourself very smart from others take the pain to ask what is the matter. Ehsan On Thu, 3 Mar 2005 14:09:06 +0200, Yair Hakak [EMAIL PROTECTED] wrote: See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ 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] Wrong CVS version ?
Robert Rozman wrote: Hi, I've updated my Asterisk 3 times with : cvs checkout -r v1-0 zaptel asterisk asterisk-addons and then do cd asterisk make clean make make install make samples make progdocs and then when I run Asterisk I get : Asterisk CVS-v1-0-02/11/05-01:46:25, Copyright (C) 1999-2004 Digium. Is this a bug in CVS handling or am I doing something wrong ? How to check which version of CVS I have from commandline ? If you're updating use cvs update -r v1-0 zaptel asterisk asterisk-addons The version printed comes from the .verison file in Asterisk. Remove it and rebuiild. ___ 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] country/city codes
On Thu, Mar 03, 2005 at 06:25:09AM -0800, VoIP Services wrote: Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 Does anyone know a formula for determining which part of a dialled number is the country code and city code ? There is no formula, you need to make a list. There are lists around the place, telling you what each prefix means. It also changes over time, as countries change their numbering to deal with growth of population and services. http://www.wtng.info has a lot of useful information... Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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] country/city codes
VoIP Services [EMAIL PROTECTED] wrote: [...] Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 I think you'll find that the country code for Bermuda is not 441. I'd have to find a telephone directory to check, but I bet the country code is actually 1, and 441 is the area code. Does anyone know a formula for determining which part of a dialled number is the country code and city code ? There's no formula - you need to use a look-up table. -- She's the kind of girl who climbed the ladder of success wrong by wrong. - Mae West ___ 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] RE: Getting phpconfig to work?
Julius Kidubuka wrote: When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; Try a simple php-script in this directory. Something like this, name it test.php ?PHP print (test) ? If it does not print test, and you see the code instead, check your httpd.conf for a general php-parsing problem. True, I have tried that and all I see is code instead. How do I go about solving the php-parsing problem in my httpd.conf then? Try installing mod_php. Your web server does not consider .php files to be scripts. I don't know why. ___ 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] Re : Calling card platform
I completely agree with Yair, especially considering the fact that we used to share the same work place. It is one thing to endorse a platform, it's a different thing endorsing your own platform in a coat of I'm a happy user. Dimi Telecom also provides calling card platforms and various voice enabled platforms, but we don't go telling everybody how wonderful our platforms are. If someone needs a good TCO/ROI platform, they will eventually get to find us (for some reason, they all do, even old SigValue clients). Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Thursday, March 03, 2005 4:51 PM To: M. Ehsanul Karim; asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re : Calling card platform It is fine to tout your own products. we call that marketing. However, anyone who claims that they can endorse a product and not mention that they worked for the manufacturer 5 months ago, and thinks this is an ethical thing to do, is not worth my time. Once again, i don't care about the platform. It's probably a very good platform. Your recommendation does not change anything about the platform, but it does call your integrity as a recommender into question. This conversation is over. -yair On Thu, 3 Mar 2005 20:41:55 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: I am a customer and I am paying them every month. I was giving out my personal opinions about the soft . What's so wrong with that if I had not said that I worked with this company 5 months ago ? Don't you have your eyes and judgements before you can buy the product ? So as you know I wokred with them it makes me a fraud or changes the whole software? Please tell me the impact of knowing I worked with them 5 months ago ? I think what you have done so far is not decent enough . You have the right the say anything but which are fact and you know it to be. It is something if I say you have your own platform and you are jealous to let know others about a good platform. I think all the people here are matured enough to get their judgements on the product rather than jsut ordering it because I said so. Ehsanul Karim On Thu, 3 Mar 2005 14:31:56 +0200, Yair Hakak [EMAIL PROTECTED] wrote: I will make this as clear as i possibly can. 1. i am not very smart from others. I am, however, a big fan of honesty. 2. You WERE NOT honest enough to say what you do. I don't care if you were or are a freelancer, or the CEO, or if they paid you in cows instead of money. You have or had a relationship with the company. You did not mention this. In fact, you painted yourself as a customer, which you might be, but it's not exactly an unbiased recommendation if you used to work there. 3. You can flame me all you want. Evidently i have a fraudulent mentality so it's OK. 4. I'm taking the pain: what is the matter? Were you planning a second email to inform us of your association with the company, did you assume we just all knew you had such an association, did you think it's not relevant? 5. i want to make it clear I know absolutely nothing about this product. It might be really great. Please keep in mind that my anger is at the poster and not the company. -yair On Thu, 3 Mar 2005 18:21:40 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: Yair, I am honest enough to say what I do , Don't jump into something you don't know...I was working there for a while and that was months ago and it was a part of my freelance contribution. Don't think others to have same kind fruadelent mentality that you have.SO next time before proving yourself very smart from others take the pain to ask what is the matter. Ehsan On Thu, 3 Mar 2005 14:09:06 +0200, Yair Hakak [EMAIL PROTECTED] wrote: See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
[Asterisk-Users] Detect sound and continue, like BackgroundDetect() for voice
I'm looking for an application that can monitor a channel for voice input and then proceed on. The closest thing I've found is BackgroundDetect, which expects DTMF. Here's what I'm doing: -Call file generated which calls someone and connects them to an extension. -Extension plays stuff, etc. etc. etc (not important) With digital or VoIP termination, this works fine, because * knows when the line is answered. On analog POTS, it has no idea when the call is actually answered, only when its dialed, so the playback starts right after the line is dialed, not after the called party picks up. The Dialogic IVR SDK monitors call termination status this way, so I'm looking for something similar in *. Anyone have any ideas on this one? Or am I going about this the hard way and missing an obvious alternative? Thanks, Daryl ___ 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] CDR
can you recomend me some bibliography??? wertJens Kübler [EMAIL PROTECTED] wrote: Am Donnerstag 03 März 2005 14:59 schrieb R A: I need that my records cdr only get the calls that begin with 9 or any other rule is this possible??yesJens___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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] country/city codes
Yet another example of someone who couldn't take 2 min to google: http://lists.digium.com/pipermail/asterisk-dev/2004-May/004151.html -Matthew - Original Message - From: Nir Simionovich [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 03, 2005 8:46 AM Subject: RE: [Asterisk-Users] country/city codes To my knowledge, there is no such formula. However, you can obtain a database of the entire ITU E164 numbering plan at http://www.numberingplans.com http://www.numberingplans.com/ , which have an updated database of all that information. Nir S _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Services Sent: Thursday, March 03, 2005 4:25 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] country/city codes Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 Does anyone know a formula for determining which part of a dialled number is the country code and city code ? ___ 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] Re : Calling card platform
Yair, I have been dealing with Amarfone as well as Ehsanul Karim for an year now and I never had any issue with them. Both were and have been customers. Ehsan is an honest individual. He might have omitted mentioning that he worked for Amarphone in the past. It does not make him a cheat or a fraud. I recommend IBM Servers and Citibank Checking Account as the best in their products and services, Servers and Banking. I worked for both of them in the past and I knew them. Am I a dishonest person? Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Thursday, March 03, 2005 9:51 AM To: M. Ehsanul Karim; asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re : Calling card platform It is fine to tout your own products. we call that marketing. However, anyone who claims that they can endorse a product and not mention that they worked for the manufacturer 5 months ago, and thinks this is an ethical thing to do, is not worth my time. Once again, i don't care about the platform. It's probably a very good platform. Your recommendation does not change anything about the platform, but it does call your integrity as a recommender into question. This conversation is over. -yair On Thu, 3 Mar 2005 20:41:55 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: I am a customer and I am paying them every month. I was giving out my personal opinions about the soft . What's so wrong with that if I had not said that I worked with this company 5 months ago ? Don't you have your eyes and judgements before you can buy the product ? So as you know I wokred with them it makes me a fraud or changes the whole software? Please tell me the impact of knowing I worked with them 5 months ago ? I think what you have done so far is not decent enough . You have the right the say anything but which are fact and you know it to be. It is something if I say you have your own platform and you are jealous to let know others about a good platform. I think all the people here are matured enough to get their judgements on the product rather than jsut ordering it because I said so. Ehsanul Karim On Thu, 3 Mar 2005 14:31:56 +0200, Yair Hakak [EMAIL PROTECTED] wrote: I will make this as clear as i possibly can. 1. i am not very smart from others. I am, however, a big fan of honesty. 2. You WERE NOT honest enough to say what you do. I don't care if you were or are a freelancer, or the CEO, or if they paid you in cows instead of money. You have or had a relationship with the company. You did not mention this. In fact, you painted yourself as a customer, which you might be, but it's not exactly an unbiased recommendation if you used to work there. 3. You can flame me all you want. Evidently i have a fraudulent mentality so it's OK. 4. I'm taking the pain: what is the matter? Were you planning a second email to inform us of your association with the company, did you assume we just all knew you had such an association, did you think it's not relevant? 5. i want to make it clear I know absolutely nothing about this product. It might be really great. Please keep in mind that my anger is at the poster and not the company. -yair On Thu, 3 Mar 2005 18:21:40 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: Yair, I am honest enough to say what I do , Don't jump into something you don't know...I was working there for a while and that was months ago and it was a part of my freelance contribution. Don't think others to have same kind fruadelent mentality that you have.SO next time before proving yourself very smart from others take the pain to ask what is the matter. Ehsan On Thu, 3 Mar 2005 14:09:06 +0200, Yair Hakak [EMAIL PROTECTED] wrote: See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/ 064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at [EMAIL PROTECTED] Ehsanul Karim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
Re: [Asterisk-Users] CDR
sorry i´m using MySQL database. there are somethingelse that you need to now?? wertYair Hakak [EMAIL PROTECTED] wrote: hi,you need to tell us how you're saving your cdr's - database, csv, whatever?-if you're saving to a database a stored procedure is probably best,unless you want to change the SQL statements in the proper module.yairOn Thu, 3 Mar 2005 05:59:59 -0800 (PST), R A <[EMAIL PROTECTED]>wrote: I need that my records cdr only get the calls that begin with 9 or any other rule is this possible?? thanks in advance wert Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSC RIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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] Detect sound and continue, like BackgroundDetect() for voice
Yes, I'm replying to my own post. Roger Gulbranson suggested this: http://www.voip-info.org/tiki-index.php?page=NVBackgroundDetect As he's using it for FAX detect, and it has a talk option as well. If anyone is interested, I'll report back with my results. Thanks Roger! Daryl ___ 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] IAX users in Japan or Taiwan?
Hi, would there be anybody on this list who uses IAX-based VoIP from Japan or Taiwan? Al (in Japan) -- ___ 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] country/city codes
VoIP Services wrote: Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 And some country codes are one digit, like 1 for US ___ 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] Detect sound and continue, like BackgroundDetect() for voice
Daryl G. Jurbala wrote: I'm looking for an application that can monitor a channel for voice input and then proceed on. The closest thing I've found is BackgroundDetect, which expects DTMF. Here's what I'm doing: -Call file generated which calls someone and connects them to an extension. -Extension plays stuff, etc. etc. etc (not important) With digital or VoIP termination, this works fine, because * knows when the line is answered. On analog POTS, it has no idea when the call is actually answered, only when its dialed, so the playback starts right after the line is dialed, not after the called party picks up. The Dialogic IVR SDK monitors call termination status this way, so I'm looking for something similar in *. Anyone have any ideas on this one? Or am I going about this the hard way and missing an obvious alternative? Yes, you missed an obvious option - search the mailing list. This has come up an number of times. Regards, Steve ___ 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] Recomended server hardware
I intend to replace our Lucent Index telephone system with Asterisk and need to buy a proper server to run it on. I have read about the problems with the HP DL380 G4 and the TE410P cards. I have a TE110P and will be using a TDM400 card for the backup analogue lines. Is there any server that you could recomend? It needs to have RAID support, hot swapping drive bays for reliability, and be rack mountable but apart from that it does not matter which one. Regards Gareth ___ 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] Re : Calling card platform
I do not recall calling anyone a cheat or a fraud. We have a saying where i am from, something about a burglar, and a hat, and fire. I'll leave it at that. As for your last question, i can't answer that. -yair On Thu, 3 Mar 2005 10:26:05 -0500, Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote: Yair, I have been dealing with Amarfone as well as Ehsanul Karim for an year now and I never had any issue with them. Both were and have been customers. Ehsan is an honest individual. He might have omitted mentioning that he worked for Amarphone in the past. It does not make him a cheat or a fraud. I recommend IBM Servers and Citibank Checking Account as the best in their products and services, Servers and Banking. I worked for both of them in the past and I knew them. Am I a dishonest person? Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Thursday, March 03, 2005 9:51 AM To: M. Ehsanul Karim; asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re : Calling card platform It is fine to tout your own products. we call that marketing. However, anyone who claims that they can endorse a product and not mention that they worked for the manufacturer 5 months ago, and thinks this is an ethical thing to do, is not worth my time. Once again, i don't care about the platform. It's probably a very good platform. Your recommendation does not change anything about the platform, but it does call your integrity as a recommender into question. This conversation is over. -yair On Thu, 3 Mar 2005 20:41:55 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: I am a customer and I am paying them every month. I was giving out my personal opinions about the soft . What's so wrong with that if I had not said that I worked with this company 5 months ago ? Don't you have your eyes and judgements before you can buy the product ? So as you know I wokred with them it makes me a fraud or changes the whole software? Please tell me the impact of knowing I worked with them 5 months ago ? I think what you have done so far is not decent enough . You have the right the say anything but which are fact and you know it to be. It is something if I say you have your own platform and you are jealous to let know others about a good platform. I think all the people here are matured enough to get their judgements on the product rather than jsut ordering it because I said so. Ehsanul Karim On Thu, 3 Mar 2005 14:31:56 +0200, Yair Hakak [EMAIL PROTECTED] wrote: I will make this as clear as i possibly can. 1. i am not very smart from others. I am, however, a big fan of honesty. 2. You WERE NOT honest enough to say what you do. I don't care if you were or are a freelancer, or the CEO, or if they paid you in cows instead of money. You have or had a relationship with the company. You did not mention this. In fact, you painted yourself as a customer, which you might be, but it's not exactly an unbiased recommendation if you used to work there. 3. You can flame me all you want. Evidently i have a fraudulent mentality so it's OK. 4. I'm taking the pain: what is the matter? Were you planning a second email to inform us of your association with the company, did you assume we just all knew you had such an association, did you think it's not relevant? 5. i want to make it clear I know absolutely nothing about this product. It might be really great. Please keep in mind that my anger is at the poster and not the company. -yair On Thu, 3 Mar 2005 18:21:40 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: Yair, I am honest enough to say what I do , Don't jump into something you don't know...I was working there for a while and that was months ago and it was a part of my freelance contribution. Don't think others to have same kind fruadelent mentality that you have.SO next time before proving yourself very smart from others take the pain to ask what is the matter. Ehsan On Thu, 3 Mar 2005 14:09:06 +0200, Yair Hakak [EMAIL PROTECTED] wrote: See, here's the problem when you misrepresent yourself...the web is so easy to search that any idiot like me can discover what you're doing. http://lists.digium.com/pipermail/asterisk-users/2004-September/ 064464.html 'nuff said. i'm sure their support is awesome. i'm sure it doesn't cost you a lot of money. I'm sure you're very fond of your own product. I'm also sure if you're you, then support is really awesome because you never have to worry about not getting back to yourself. -yair On Thu, 3 Mar 2005 17:46:23 +0600, M. Ehsanul Karim [EMAIL PROTECTED] wrote: We are using a platform from AmarFone Inc. It great full
Re: [Asterisk-Users] CDR
Hello, Nir's suggestion seems to be best...is there a specific reason you don't want to save certain CDR's? Better to save everything and pull out what you need when you need it. -yair On Thu, 3 Mar 2005 07:33:03 -0800 (PST), R A [EMAIL PROTECTED] wrote: sorry i´m using MySQL database. there are something else that you need to now?? wert Yair Hakak [EMAIL PROTECTED] wrote: hi, you need to tell us how you're saving your cdr's - database, csv, whatever?- if you're saving to a database a stored procedure is probably best, unless you want to change the SQL statements in the proper module. yair On Thu, 3 Mar 2005 05:59:59 -0800 (PST), R A wrote: I need that my records cdr only get the calls that begin with 9 or any other rule is this possible?? thanks in advance wert Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSC RIBE 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 Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web ___ 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] What my IAXy could have been...
Quoted Message 08:18 2005.03.03 -0600, from Matt Schulte: you and everyone else :-) How about these products from China - has anybody tried any of them? http://www.farfon.com/ http://ipphone.eezeephone.com/ http://www.iaxtalk.com/ But i AM looking for info on another IAX capable device - like the IAXy, but more user friendly, as it were... -- ___ 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] Warning Message with voicemail CVS 3-3-05
I just updated our asterisk zaptel libpri to the cvs 3-3-05 8:07am and now after leaving a voicemail we are getting the following in our logs : Mar 3 10:50:25 WARNING[4408]: Can't change device '**Unknown**' with no technology! Mar 3 10:50:25 WARNING[4409]: Can't change device '**Unknown**' with no technology! Mar 3 10:50:25 WARNING[4410]: Can't change device '**Unknown**' with no technology! No other changes to configuration were made for the update any ideas? Regards Greg ___ 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
