similar to: Manager API

Is there a way with the command (1.4.42) for sip show peers to see the FULL "Name/Username" field??? I have long names and mine are being truncated. Thanks Jerry

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

Since www.bkw.org seems not to exist anymore (getting response from some hosting provider), does anyone happend to have a copy of app_valetparking.c from www.bkw.org - the one that should work with * stable 1.0.X ? If so please contact me. One that can be downloaded from www.loligo.com dosn't compile with 1.0.X, and SuperValletParking (www.asterlink.com/svp/) seems to be for * HEAD

Nope ! This is the one that tries to include PRE 1.0.X header file <parking.h>. It cannot compile on * 1.0.X (I have tried also to include <features.h> instead of <parking.h> (as far as I know features.h is successor to parking.h), but still without results). Thanks anyway. Nenad > > Try this > >> Since www.bkw.org seems not to exist anymore (getting

Hello, With an extensions.ael enabled system, I keep getting whatever I change into my "astup.call" file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

Hello, Is there or can anyone provide a comprehensive guide (designed for Linux/Asterisk novices) to installing/setting up Asterisk in order to support Fax2Mail service? In my case, I would like Asterisk to receive fax calls to predefined numbers (ranges) and to associate each of these numbers to email addresses. Thank you in advance. David --------------------------------- Yahoo!

In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very short (0-1s) duration for the CDR that results from this call, regardless of the time spent running the FastAGI application. I want the CDR

Hi, I can't make outgoing calls via Broadvoice. I have tried each and every configuration that was posted to list previously. I am able to receive incoming calls fine. I get the following in asterisk console: ===================================================== asterisk*CLI> show version Asterisk CVS-HEAD-03/10/05-22:51:28 built by vicky@asterisk on a i686 running Linux

I've been banging my head on a brick wall for about an hour now trying to understand why the following doesn't work (which is even provided as an example in the distribution!). The goal is to create a voicemail-only extension not associated with a phone. I'd rather not have an extension dedicated to VoicemailMain(), so I would like the user to be able to hit '*' during

Hi ALL; I got the latest Asterisk-addons for Mysql-Cdr, but I have problem compiling that.It says: # make ..... ........ res_config_mysql.c: In function `realtime_mysql': res_config_mysql.c:143: warning: passing arg 1 of `ast_strlen_zero' makes pointer from integer without a cast res_config_mysql.c: In function `realtime_multi_mysql': res_config_mysql.c:242: warning: passing arg

I upgraded to the latest stable version of 1.0 today and am still seeing the *8 problem where the phone that was originally dialed keeps on ringing even after another phone picks up. Are other people also seeing this? Has somebody figured out how to make this go away? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I need help with modifying cdr_addon_mysql.c I want to have more fields in cdr table in asterisk. I've tried to modify cdr_addon_mysql.c and replace userfield with ex team (sed -e 's/userfield/team/g' ). When I try to recomplie menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'.

Hi, Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some events but sometimes not. I think Linksys PAP-2 has a problem with recognizing digits in appropriate

there is a problem in compiling asterisk-addons any one have fixed this problem. i want res_config_mysql.so any one help me ----------------------------------------------------- [root@localhost asterisk-addons]# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o res_config_mysql.o res_config_mysql.c res_config_mysql.c: In function `realtime_mysql':

Hi, I have to build Asterisk server for about 30 user (30 concurrent calls). I decided to buy this box: -- motherboard Intel E7210 + Hence Rapids -- processor P4 3.0 GHz -- RAM 2x512 MB DDR ECC -- network interface Intel 82541 GI Is this configuration enough to handle 30 users at the same time. I am not planning to use any transcoding (everything will be alaw). Cheers Andrew --------------

Hi, Few days ago I installed Email2fax application on my Asterisk box. I works but not in 100 %. Sometimes (to be certain quite often) I don't receive the whole fax. My fax machine cuts off transmission in 1/2 or 1/3 of the page. I read about it on a wiki and some user lists and people say that this behaviour could be cause be the Ghostscript and the conversion to the tiff format, but when I

iaxComm is a cross-platform IAX2 softphone available for Win32 and Linux. Win32 and Linux binaries as well as the LGPL source are available at: http://iaxclient.sourceforge.net Recent improvements are a less cluttered user interface, audible ringback and audible outgoing ring, and of course IAX2 protocol support. iaxComm is based upon the wxWindow GUI framework and compiles on Microsoft

Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX at Inside-1f32;1 setting write format to slin from alaw native formats (alaw) What can I do to lose it. I am using asterisk 10.1.2 Best regards

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch <http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch