similar to: Asterisk died - exactly every 60 minutes

You may loose 'control' of the call but you can always 'get it back' Use the UnigueID of the call to track it throught Asterisk. You can palce a monitor event to redirect, bridge, drop, answer or antything else. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Sebastian

Hello, I try to compile the driver zaptel and they give the following error: linux01:/usr/src/zaptel# make install gcc -Iir/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -Iir/drivers/ l -I. -Wstrict-prototypes -fomit-frame-pointer -Iir/drivers/net/wan -Iir /net -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c In file included from zaptel.c:44: /usr/include/linux/module.h:21:

Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference "FM" is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad

Hi, I am looking for a softphone (sip or iax) that works in Solaris/SPARC with sunray100 terminals. I found iaxcomm but it doesn't work. Also I am trying sip-communicator but I have several errors from JMF/RTP. Does anyone have a softphone working over this platform? which one? I don't care if it is a commercial product, I can buy it if works fine. thanks in advance. Sebas --

Should be mailbox = 7752365815@vpbx-wpti Best Regards, Ben Bawkon --------- Original Message --------- From: Bruce Komito To: <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] voicemail notify problem on sip extension Sent: 5/19/2004 4:27:51 PM I'm having a problem with the voicemail notify feature. Although I have the voicemail box configured for the sip extension, the

I try to change something in ASTCC, but I am now totally blind, .... I hang on one line now. I changed: vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi 22c22 < # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN}) --- > # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${TARIFF},${EXTEN}) 35c35 < # exten =>

vpbx*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 621/621 192.168.250.76 D N 5060 OK (65 ms) 626/626 192.168.250.109 D N 5060 OK (180 ms) 616/Ronald Softphone (Unspecified) D N 0 UNKNOWN 615/Ronald office 192.168.250.103 D N 5060 OK (41

I have an * system running version 1.0.8 and it works mostly fine. I am using it as a virtual PBX and we share the box among companies. I have the calls all staying separate, we well as the companies' extensions, voicemail, etc. The only problem I'm having is with two accounts that use the same SIP termination provider. * seems to be confusing the sip contexts for the incoming calls.

Hi i wanna know exactly the principal functions of ldap, if is posible send me a example because im not very clear about this protocol with samba. thanks in advance Cristhian Dominican Republic -- "No basta saber, se debe tambi?n aplicar. No es suficiente querer, se debe tambi?n hacer"(Goethe, Johann Wolfgang von). Saludos Cristhian Nunez AGB-CDI Dominicana Abraham Lincoln

I tried to add the databases from the localhost to the database server and changed the every /etc/asterisk/*.conf from host=localhost to host=192.168.10.10 (my dababase server) When I restart asterisk, I do not get any errors, but after a phone call I see: Jun 7 18:11:56 ERROR[7877]: cdr_addon_mysql.c:400 my_load_module: Failed to connect to mysql database cdr on 192.168.10.10 Or if I try

Hi all, We just set up our new mail servers using Postfix and Dovecot. We have implemented a load balanced solution to where our two mail servers have a common NFS share. Everything is working great. Everything but Dovecot for the most part. In testing, Dovecot was great. Now that we're in the production environment, we've noticed that every 20 minutes, Dovecot will stop

I tried to install astguiclient and it gives me for each follow page: Object not found! Looking into the apache log file I find: [Sun Feb 27 16:18:30 2005] [error] [client 192.168.250.108] File does not exist: /srv/www/htdocs/astguiclient/method=POST, referer: http://vpbx.elmit.com/astguiclient/phone_stats.php?extension=gs102&server_ip=192.168.250.20 I have uncommented the ### If you have

Hi, all. I've been searching google, bug reports and forums and have looked in all the asterisk-users list archives back to 2003 but haven't seen an answer to this, so thought I'd post here. The problem seems to be that Asterisk 1.6.0.5 is sending backslashes (needed to escape commas and so forth in 1.4.21.2) as *literal* backslashes to Mysql, so that Mysql gives a syntax error

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call

Hello, I used a mac G5, R.2.1.1, and G77 3.4.4 and I would like to use and call a fortran subroutine. The trouble is that it seems I am not able to correctly load the compiled code. Here is what I have done: In the terminal this how I compiled my fortran code: R CMD SHLIB ~/Desktop/Fortan_kmeans/kmeans3.f There is the wrapper I have paste inside de kmeans3.f file: c

I try to install astcc. Make install gives me: DBI connect('database=astcc;host=localhost','astcc',...) failed: Unknown database 'astcc' at ./astcc-admin.cgi line 67 DBI connect('database=;host=','',...) failed: Access denied for user: 'root@localhost' (Using password: NO) at ./astcc-admin.cgi line 60 Ignoring that (!!! ;-) ) and going to the

Below is my extensions.conf for the fax part [incoming_28345474] ; ;******************************************************************** ; BEGIN - Inbound call handlers ;******************************************************************** ; exten => 8862100,1,NoOp(${CALLERID(num)}) exten => 8862100,2,Background(if-u-know-ext-dial) exten =>

I got a new motherboard and upgraded Asterisk, ... However, I am not lucky enough to get it running again. To compile went ok, besides this remark at the end of asterisk: WARNING WARNING WARNING Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with

Hi, I have Mandriva 2006 x86_64. I tried to install wine, but the binaries are for 32 bits and the result is very instable (win programs close suddenly, no internet available for those programs, etc...) So I tried to compile wine 0.9.14 from source, but I have an error during make when "searching for -lXext" : LD_LIBRARY_PATH="../../libs/unicode:$LD_LIBRARY_PATH"