Re: [Asterisk-Users] Multiple Servers and 1 Central Voicemail
Hi all, what about 1) link the /etc/asterisk/voicemail.conf to /mnt/nfs/asterisk/etc/asterisk/voicemail.conf 2) link /var/spool/asterisk/voicemail to /mnt/nfs/asterisk/var/spool/asterisk/voicemail It works fine for me. BTW: if you chroot asterisk daemon remember to set the user UID to the right one on each asterisk server. Josiah Bryan wrote: On Tuesday 12 April 2005 11:49 am, Luki wrote: Also, what happens if for example, the user is accessing his VMB on server 1 and changes his password, then travel to where server 2 is and tries to access his VMB? the config on server2 would still have the old one so you need to sync voicemail.conf on all servers too ... If you use the realtime config via a DB, it should be OK. But I still don't think that MWI will work properly if a message is left on server A and user is actually registered on server B, which is NOT on the same network and hence does NOT share the same voice mail spool. How will B know there is a message left on A for the same user? Does realtime share this info too? And if so, how does the message get retrieved if B does not have access to files on server A, where the actual message is? Why not just NFS mount the /var/spool/asterisk/voicemail directory from a central server? That way, all servers share the same spool and the MWI will get reflected on all servers. -josiah ___ 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] Multiple CDR Locations
Hi Aaron, you can simply add the cdr library u need in the default /usr/lib/asterisk/modules. I don't think u can load the same module twice but u can load cvs ad pgsql module at the same time. U have to change the Makefile inside the /cdr to build the module u need. Bye Aaron Daniel wrote: Does anyone know of a way to have asterisk save multiple cdr records in different places (i.e. the same record in a database locally and in another database on another system, or database and csv, or some other strange combination)? Aaron 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
[Asterisk-Users] Optional URL param
Hi all, I'm looking for a IAX/SIP client that supports the url param of the Dial/Queue command. Thanks in advance. ___ 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: Spandsp loading via asterisk app_rxfax.c brokenpipe.
It should be a mpg123 problem, not a spandsp problem. Stop asterisk, make clean, make install and start asterisk again. Have fun. "Ariel Batista" <[EMAIL PROTECTED]> wrote in message news:<[EMAIL PROTECTED]>... I have compiled Spandsp without any problems. I got no errors I have also done the patch without getting any error. I have tried pre4 and pre6 version with same problem. I got no errors with the ./configure make make install. After I add the directory for /var/local/lib into the /etc/ld.so.conf I have ran ldconfig. I am running asterisk as user asterisk on a RH 9 Linux distro. This system has been running fine for months. I am also using Asterisk stable version 1.0.3 from 12/07/04. But when I start asterisk I get Broken Pipe notice and asterisk does not start. If I put into the module.conf noload = apps_rxfax.so and app_txfax.so I am able to load asterisk fine. But as you known with out the other files loaded I can't use the spandsp fax function. I don't know where to start to trouble shoot. I have read the FAQ on opencall.org but could not find what might be my problem. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 -> SPANDSP
Hi all, i'm looking for docs about a stable configuration of spandsp that works whit *. I use asterisk CVS-v1-0-11/19/04-19:26:32, libtiff-3.5.7-11, libtiff-devel-3.5.7-11, spandsp 0.0.1k, audiofile-0.2.3-6 and audiofile-devel-0.2.3-6, but every time i use rxfax asterisk hungs and an 8 k fax file is written Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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.conf extensions.conf
Hi Mauro, can u send me yours sip.conf and extension.conf ? Ciao Gianluca - Messaggio originale - >Da: "Mauro Locatelli"<[EMAIL PROTECTED]> >Inviato: 05/11/04 16.47.23 >A: "Asterisk Users Mailing List - Non-Commercial Discussion"<[EMAIL PROTECTED]> >Oggetto: [Asterisk-Users] sip.conf extensions.conf >I have an asterisk server(x100p wildcard) that function as a gateway. >I have some local soft phone (for example 3) and I want: > >1- call from one internal softphone the other internal softphone >2- call out on the PSTN from internal softphone >3- call out on the sipphone.com >4- receive call from external PSTN and choose wich internal softphone ring >5- receive call from sipphone and choose wich internal softphone ring > >I make sip.conf and extensions.conf but only 1,2,3 point work.. > >If someone is in my situation, and work all full, can send me his sip.conf and extensions.conf for compare? > >Very very thanks and sorry for english.. >Mauro > > [Messaggio troncato. Toccare Modifica->Segna per il download per recuperare la restante parte.] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 and Eicon Diva 2.0
Hi all. This is my first post, but i don't know if i'm off topic. I've a linux box with RedHat 9.0 (linux kernel 2.4.8-20) and asterisk cvs installed on it, Eicon Diva 2.0 and a BRI ISDN line in italy. Everything works fine but i can't handle the CALLER NUMBER: it's alway set to 0. Can someone in Italy help me ? Thanks in adcance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ISDN Card Recommendation
Hi Mark, I've succesfully installed an Eicon Diva 2.0 PCI in 10 minutes. It works fine but i have some problems with CALLER NUMBER: it's always 0. My box is RedHat 9.0 and asterisk cvs 25-08-2004. Bye -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paterson, Mark Sent: giovedì 26 agosto 2004 15.26 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ISDN Card Recommendation I'm running Asterisk 1.0 RC2 on a RedHat 9.0 box. I have a ISDN BRI line that I would like hook up to my Asterisk server and would like to ask the group what you guys would recommend as far as isdn cards that install easily into the Linux and asterisk environment. Rgs, Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
