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
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
FW: [Asterisk-Users] SIP echo on LAN
-Original Message- From: Nic le Roux [mailto:[EMAIL PROTECTED] Sent: 24 February 2005 12:39 PM To: 'Julian J. M.'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP echo on LAN Do you mean that I need to check the sound card settings on the machine that I'm dialling from or too, or on the asterisk server ? Where could I change the PCM settings ? I've been looking around but cannot find what your talking off. Any help appreciated. Rgds Nic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M. Sent: 21 February 2005 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP echo on LAN Check your soundcard controls... maybe it's recording what you hear or PCM, thus sending it again to the other party. Julianjm. On Mon, 21 Feb 2005 09:47:55 +0200, Nic le Roux [EMAIL PROTECTED] wrote: Good Morning, I have a weird situation, I'm testing with Xlite as SIP phone (is it any good ) and dialing an extension (also Xlite on same LAN) and I'm getting a real bad echo on the dialer's side and a not so bad one on the receivers side. Has anyone had something like this ? Aparently one should only get echo when you break out onto a telco network ? ___ 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] SIP echo on LAN
Hi There, Thanks for your reply. Where can I read up on doing this or maybe you could point me in the right direction. I don't believe that I have recording enabled. | Julian Wrote: | | Check your soundcard controls... maybe it's recording what you hear | or PCM, thus sending it again to the other party. | | Julianjm. | | Nic le Roux wrote: Good Morning, I have a weird situation, I'm testing with Xlite as SIP phone (is it any good ) and dialing an extension (also Xlite on same LAN) and I'm getting a real bad echo on the dialer's side and a not so bad one on the receivers side. Has anyone had something like this ? Aparently one should only get echo when you break out onto a telco network ___ 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] SIP echo on LAN
Good Morning, I have a weird situation, I'm testing with Xlite as SIP phone (is it any good ) and dialing an extension (also Xlite on same LAN) and I'm getting a real bad echo on the dialer's side and a not so bad one on the receivers side. Has anyone had something like this ? Aparently one should only get echo when you break out onto a telco network ? ___ 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] Problems compiling on mandrake
I have it installed and working 100% on Mandrake 10.1 Maybe missing development libs are the cause. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: 18 February 2005 09:53 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Problems compiling on mandrake Guys.. Im having problems compiling asterisk under Mandrake 10.1 with kernel 2.6.8.1-12mdk The error I get is: In file included from chan_phone.c:36: /usr/include/linux/ixjuser.h:353: error: syntax error before '*' token make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/root/software/asterisk-1.0.0/channels' make: *** [subdirs] Error 1 Any ideas what might be wrong? __ Anton Krall ___ 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] Eicon Diva audio problem [Newbie]
Hi
all,
I have trouble
getting my setup configured properly.
I have a Eicon|DIVA
Server BRI-2M/-2F card installed, using melware driver and following asterisk
wiki guidelines.
However whe I try to
dialup the number I get only silence and after a while
disconnection.
The following is
displayed on the console.
What am I doing
wrong ?
*CLI == Starting
CAPI[contr1/0991]/0 at demo,0991,1 failed so falling back to exten
's' -- Executing Wait("CAPI[contr1/0991]/0", "1") in new
stack -- started pbx on channel (callgroup=2)!
== Spawn extension (demo, s, 1) exited non-zero on
'CAPI[contr1/0991]/0'
Thanks and
Kind Regards
Nic
___
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] Audio Quality over LAN very bad
Hi All, I'm running Asterisk on the following vendor_id : GenuineIntelmodel name : Celeron (Coppermine)cpu MHz : 668.202cache size : 128 KB with 192 MB Ram Audio coming from Asterisk (the demo ) is excellent when using a SIP phone on the LAN to Asterisk, and when dialling in from outside via ISDN to Asterisk. However, when connecting from SIP phone to SIP phone (across LAN) and dialling from externally to SIPwhich is on the local LAN it is very choppy and one can barely make out the other party. I'm using an Eicon Diva 2-m card and 100mb network all round. What could be the cause as I believe bandwidth is ruled out. Thanks and regards Nic ___ 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] Audio Quality over LAN very bad
Thanks for the reply,
It was on GSM, Ive changed to ulaw last
night,
It did make a differance but I'd say its still not as good
in quality as the recorded messages being played back.
What is the suggested or should I say, "Best Practise" when
it comes to audio codecs used on asterisk ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chamberland-Larose, GuillaumeSent: 01 February 2005 03:07
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: RE: [Asterisk-Users] Audio Quality over LAN very
bad
Maybe you're transcoding on the server with cpu intensive
codecs? That would be the first thing I'd look at. Try using G.711
(ulaw)on both SIP phones and remove reinvite=no and canreinvite=no from
your phone declarations in sip.conf.
Hope that helps.
Guills
From: Nic le Roux
[mailto:[EMAIL PROTECTED] Sent: Monday, January 31, 2005 7:01
AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Audio Quality over LAN very bad
Hi
All,
I'm running
Asterisk on the following
vendor_id : GenuineIntelmodel
name : Celeron (Coppermine)cpu
MHz : 668.202cache
size : 128 KB
with 192 MB Ram
Audio coming
from Asterisk (the demo ) is excellent when using a SIP phone on the LAN to
Asterisk,
and when
dialling in from outside via ISDN to Asterisk.
However, when
connecting from SIP phone to SIP phone (across LAN) and dialling from
externally to SIPwhich is on the local LAN
it is very
choppy and one can barely make out the other party.
I'm using an
Eicon Diva 2-m card and 100mb network all round.
What could be
the cause as I believe bandwidth is ruled out.
Thanks and
regards
Nic
___
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 1.0.3 startup
Hi All, I've managed to compile make and make install asterisk on Mandrake 9.2. However on startup I get the following message: [cdr_tds.so]Jan 20 11:13:54 WARNING[20999]: loader.c:258 ast_load_resource: libtds.so.3: cannot open shared object file: No such file or directoryJan 20 11:13:54 WARNING[20999]: loader.c:440 load_modules: Loading module cdr_tds.so failed! I have freetds installed from RPM, and the lib is here /usr/local/lib/libtds.so.3.0.0 Where does asterisk look for the lib ? Maybe I can do a symlink ? Any help appreciated. Thanks and Regards Nic ___ 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] FW: Asterisk 1.0.3 startup
Sorry all, Did that and its going good now. Rgds Nic From: Nic le Roux [mailto:[EMAIL PROTECTED] Sent: 20 January 2005 11:22 AMTo: 'asterisk-users@lists.digium.com'Subject: Asterisk 1.0.3 startup Hi All, I've managed to compile make and make install asterisk on Mandrake 9.2. However on startup I get the following message: [cdr_tds.so]Jan 20 11:13:54 WARNING[20999]: loader.c:258 ast_load_resource: libtds.so.3: cannot open shared object file: No such file or directoryJan 20 11:13:54 WARNING[20999]: loader.c:440 load_modules: Loading module cdr_tds.so failed! I have freetds installed from RPM, and the lib is here /usr/local/lib/libtds.so.3.0.0 Where does asterisk look for the lib ? Maybe I can do a symlink ? Any help appreciated. Thanks and Regards Nic ___ 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
