similar to: asterisk-users Digest, Vol 52, Issue 81

*Good morning! * *I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2, I verified in the repository that has to libraries of the project openR2, but I don't manage to do to work in the trixbox, when I type the command (it colors show channeltypes)ele no demostra the support to MFC+R2, they could help finding out which package is necessary of the trixbox and which the

Hi, I use Asterisk-1.2.26 (with Trixbox-2.1.12) and Portech MV-370 and my problem is that when I try to call an external mobile phone via Portech I have alway busy and in log file: Called Portech/348xxxxxxx -- Got SIP response 486 "Busy Here" back from 192.168.1.2-- SIP/Portech-086e5ee0 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing

i currently have a samba pdc, samba bdc and samba member server all running samba-3.0.23c-1.fc5. up until the 3.0.22 releases, i never had any problems with users authenticating to member servers. problem now is, a user from windows xp professional (which is part of the domain) can auth to the pdc and bdc, but not to the domain member server. the same thing happens from windows xp home (even

On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: > I want to change call files, which has caller id in them, to call > originate from dial plan. > But I don't see such parameter here > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate > > How can I pass callerid to following: > > exten => 6003,n,Originate(SIP/6003 at

On Thursday, December 25, 2014 03:53:44 PM Dmitry Melekhov wrote: > 25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: > I want to change call files, which has caller id in them, to call > originate from dial plan. > But I don't see such parameter here >

Is there a variable for the filename that is created by the FollowMe application when "a" is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next

Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb12cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

Working with asterisk 1.4; using the AMI Originate command, it is possible to do something like: Variable: CDR(accountcode)123456 Or must the variable names be "var[n]" where n is a number? I'd like to set the accountcode for a Local channel that originates a call. Thanks. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE

with newest Asterisk version.=20 When holidays will end more and more people will start to complain about = this. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -----Original Message----- From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anthony = Messina Sent: Sunday, December 30, 2007

After updating to 1.4.15, I have the following issue: When I try to use the "M" macro option in the Dial() option, I get the following in the console: -- Executing Dial("Zap/1-1", "Zap/g2/w5051234|60|M(set-userfield^local)KT") -- Called g2/w5051234 -- Zap/3-1 answered Zap/1-1 [Dec 6 12:10:58] ERROR[19496]: app_dial.c:1541 dial_exec_full: Unable to start

After upgrading to 1.8.3.2 today, I notice that my Aastra 480i SIP phones no longer initiate hold music when a call is placed on hold. I seem to be having the same issue as the person here: http://forums.digium.com/viewtopic.php?f=1&t=77553 Has anyone else run into this issue? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC

I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work "as expected." I would like to be able to do something like: exten => _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal "*" followed by

I am a fairly novice Asterisk 1.4 user who used to use CallWeaver, based on asterisk 1.2. I used Realtime MySQL with CallWeaver and am currently using the very same MYSQL tables (and columns) with Asterisk 1.4.11 and things are working well. The questions I have are, since new configuration variables have been added into Asterisk 1.4, can I simply add columns in my MySQL sippeers table for

Hello ! I am experiencing some problems with Asterisk trunking, this is the scenario: There are 3 servers, a DID server provider (VOIP provider) which delegates us a bunch of DID numbers to our asterisk server number one (I will call it AA), from which I route the calls to Asterisk server number 2 (I will call it BB), which then terminate on phone handsets. The trouble is, that I probably

Guys, when compiling libunicall i have these errors. Any idea? creating libunicall.la (cd .libs && rm -f libunicall.la && ln -s ../libunicall.la libunicall.la) if gcc -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -MT testcall.o -MD -MP -MF ".deps/testcall.Tpo" -c -o testcall.o testcall.c; \ then mv -f ".deps/testcall.Tpo" ".deps/testcall.Po"; else rm -f

libspandsp.so.0: cannot open shared object file: No such file or directory Created the symlink: /usr/local/lib# ls -lt lib* lrwxrwxrwx 1 root staff 19 2008-11-28 22:42 libspandsp.so.0 -> libspandsp.so.1.0.0 -rw-r--r-- 1 root staff 1849266 2008-11-13 13:26 libspandsp.a -rwxr-xr-x 1 root staff 865 2008-11-13 13:26 libspandsp.la lrwxrwxrwx 1 root staff 19 2008-11-13 13:26

Make sure that /usr/local/lib is in your lib path. /etc/ld.conf ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David C. Nicosia Sent: Sunday, January 08, 2006 5:25 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] spandsp for 1.2.1 - libspandsp.so.0: cannot openshared object file:

I am getting the following error when starting: loader.c:325 __load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory loader.c:499 load_modules: Loading module app_txfax.so failed! When I load app_txfax.so and/or app_txfax.so; if these are commented out, asterisk starts fine. I downloaded the spandsp version from

Recently, I've been having issues with the URIs returned from e.164.org and toll free calls. It seems that the URIs that are returned from ENUMQUERY and ENUMRESULT are no longer the proper numbering schemes that the poviders use. I've been using the following [enum] template in my outbound route for quite some time with great success until recently. [enum](!) exten =>