Re: [asterisk-users] error when open a2billing web page!
2009/12/29 Zhang Shukun bit...@gmail.com: OK. Thanks 2009/12/29 ram talk2...@gmail.com: On Mon, Dec 28, 2009 at 10:39 PM, Zhang Shukun bit...@gmail.com wrote: hi, i have installed a2billing , when i open /admin web pages. errors as follow: Fatal error: Call to undefined function bindtextdomain() in /usr/local/src/a2billing/common/lib/languageSettings.php on line 130 One time I had the same problem and others problem like that, but was dependence problems. I don't remember right now which one of the dependence you have to install. Google it and you will find the answer. good luck, do you know what's wrong? you get quick responce if you post the same in a2bill forum look at their site forum Ram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Documentation DID + Asterisk
Hello there!!, Am looking for a manual or documentation that explain how to buy a DID number and how to configure it with Asterisk, and when some body call to that DID number Asterisk answer with a automatic operator Some body know about this manual? I already search in the web but nothing yet... Regards, Abel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] noise in Asterisk 1.4 and 1.6 versions
I had installed Asterisk 1.4 and when I call to a exist extension, the voice have noise, but, when I call to a extension does no exist, asterisk played a voice that say me that extension does no exist, but without noise I want I some body can test with a softphone my server, ip: 75.74.115.209 user: ramses pass: ramses the extension 1000 exist, try what ever other extension does not exist to hear the difference.. thanks for All, Abel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wich gateway is much better?
Hello everybody, I have a doubt If I want to send every call from a server asterisk to a gateway to a line PSTN, in the gateway what type of port I need FXO o FXS? I need to know wich gateway to buy, with port FXS or FXO? regards, Abel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk dedicated server
Hello every body I am looking for a website that provided hosted servers with asterisk and some kind a billing system... some one know about this? ABel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk on freebsd
Hello, I want to know if some body use Asterisk on Freebsd 7.0 release? My problem is that, when I call to any extension and the asterisk need to reproduced a file GSM o MP3, whatever, that have a lot of noise... Only not have noise when the extensions is not avalaible. That happen only in Freebsd 7.0, on Windows 32 don't happend that, Debian either... so, some body know why is that? Thanks, Abel Monzon___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on Freebsd 7.0 Release.
Hello, I want to know if some body use Asterisk on Freebsd 7.0 release? My problem is that, when I call to any extension and the asterisk need to reproduced a file GSM o MP3, whatever, that have a lot of noise... Only not have noise when the extensions is not avalaible. That happen only in Freebsd 7.0, on Windows 32 don't happend that, Debian either... so, some body know why is that? Thanks, Abel Monzon___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bug in Asterisk 1.4.22?
Hello is my idea or this is a bug? The thing is that I have in my asterisk.conf this: [directories] astetcdir = /usr/local/etc/asterisk astmoddir = /usr/local/lib/asterisk/modules astvarlibdir = /usr/local/share/asterisk astdatadir = /usr/local/share/asterisk astagidir = /usr/local/share/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk where the dir of agi-bin is in /usr/local/share/asterisk/agi-bin and inside agi-bin directory I have a file called a2billing.php and in my extesions.conf i have: [a2billing] exten = 1,1,answer exten = 1,2,Wait,2 exten = 1,3,DeadAgi,a2billing.php exten = 1,4,Wait,2 exten = 1,5,Hangup and then in my softphone I call to 1 the asterisk log say this: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory -- Executing [EMAIL PROTECTED]:4] Wait(SIP/abel-28c18000, 2) in new stack == Spawn extension (default, 1, 4) exited non-zero on 'SIP/abel-28c18000' So, i change the file a2billing.php to another place and I change this new place in asterisk.conf: [directories] astetcdir = /usr/local/etc/asterisk astmoddir = /usr/local/lib/asterisk/modules astvarlibdir = /usr/local/share/asterisk astdatadir = /usr/local/share/asterisk astagidir = /new/place/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk I reload the asterisk server and the asterisk log still say me the same place before: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory -- Executing [EMAIL PROTECTED]:4] Wait(SIP/abel-28c18000, 2) in new stack == Spawn extension (default, 1, 4) exited non-zero on 'SIP/abel-28c18000' Why is that? Any suggest? Thanks for all, Abel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with my softphone
Hello, when with my client X-lite try to register in the server that say me, Registration error:501 Not implemented. What isn't implemented? the registration in the sip.conf or extensions.conf? how can i implemented that? thanks. Abel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with my softphone
Hello, when with my client X-lite try to register in the server that say me, Registration error:501 Not implemented. What isn't implemented? the registration in the sip.conf or extensions.conf? how can i implemented that? thanks. Abel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio Files
Hello there, I wan to know what is the files that have the control of the quality the sound, When I call a extension, and reproduced a file gsm, or I tolk why another extension, have noise... I thinks that is because have bad quality in the .conf. Thanks. Abel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio Files
- Original Message - From: Julien Claassen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 26, 2008 8:03 PM Subject: Re: [asterisk-users] Audio Files Hi! I think all - at least all PSTN - calls have the same quality in means of bitrate, number of channels and samplerate. It's 8kHz, 16bit and mono. About noise, I didn't have problems with that. Seems it's not really about quality. Probably it would be helpful, if you tell us, which extensions/protocol you used. Kindest regards Julien Well, I had installed the sample with gmake, and I add my own extension, exten = 269544,1,dial(Sip/user1,20) exten = 269544,2,hangup() and exten = 269544,1,dial(Sip/user2,20) exten = 269544,2,hangup() exten = 1,1,Playback(Wellcome) exten = 1,2,hangup() So, When I call from user1 to user2, have noise, If I call from user1/user2 to extension 1 the Playback have noise to. but, If I call to inexitent extension like the asterisk reproduced a error sound and not have noise.. What's is wrong?? Abel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
