similar to: Asterisk svn starting problem

Hi. I have the next configuration... I dial from my analog phone in the TDM400P to extension 102, and the second agi works about 1 out of 10 times, the other nine it gives me these error on the asterisk console: -- Starting simple switch on 'Zap/2-1' -- Executing Macro("Zap/2-1", "receivecall") in new stack -- Executing AGI("Zap/2-1",

We recently had our PRI installed, we currently have 100 toll-free's pointing to it. I have almost everything working great but.. I have setup the first few numbers we want to use coming in from the PRI and they work great, but.. What I want to do is setup an extension with pattern matching to answer for any numbers called that are pointed to our system and PRI but not yet in

Hello all, I try to use a agi script to get a variable from * und put them into a script which gives me another variablke and put this in *. My problem is now it seems the var ID is empty coz i always jump into the result 0 loop. The $MSN should be in the SetCIDNum. #!/usr/bin/php -q <?php include("/var/lib/asterisk/agi-bin/phpagi.php"); $agi = new AGI(); $ID =

Hi, i can't compile the latest svn update from zaptel: /lib/modules/2.6.14-1.1653_FC4smp/build make -C /lib/modules/2.6.14-1.1653_FC4smp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.14-1.1653_FC4-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:6193:5: warning: "CONFIG_ZAPATA_DEBUG" is not defined

Dear asterisk users, I am presently playing with an asterisk@home. I am trying to find the best codec solution for my voicemail records. I want to use ARI (Asterisk Recording Interface) to read the messages. I first used the default wav encoding that was not appropriate because my navigator does not handle wav mime types correctly and I have difficulties playing wav files in my basic linux

My number is not submitted. I updated my asterisk but this error still occurs coz of the "" in the SetCallerID tag thats why it will be a empty SetCallerID is submitted. Is there a fix to correct this error? -- Executing SetCIDNum("SIP/31-752a", "4989427xxxx") in new stack -- Executing SetCIDName("SIP/31-752a", "4989427xxxx") in new stack

I have a problem with the cdr. We terminate through a pstn provider to the pstn network. The problem is now the cdr accounts the connection to the gateway. Coz the gateway is answering our call and then forward to the pstn number. So i have billsecs all the time even it is only ringing or so. Somebody has a solution for that? -- Executing Dial("SIP/1000114-fcf8",

Hello I patche dmy 7970 with the current SIP image i have 2 lines on it via sip and 6 hint speeddials but it seems thats only a speeddial no infos about busy status or so comes to the speddial button. somebody can help me?

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

Friends, Beginning next week, I will travel around Europe to teach Asterisk - the one day Meet Asterisk training. MeetAsterisk is organized by Edvina in cooperation with Digium and Voop. In many places, local Asterisk equipment resellers participate and show their equipment. This is the tour plan: * Amsterdam April 26 * Copenhagen April 27 * Oslo April 28 * Paris May 3 * Brussels May 4 *

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Is that the only problem to use lld to link cygwin programs? On Thu, Feb 8, 2018 at 8:19 AM, Andrew Kelley <superjoe30 at gmail.com> wrote: > Here are the linker errors: > > lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: > __data_start__ > lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: > __data_end__ > lld: warning:

Here are the linker errors: lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __data_start__ lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __data_end__ lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __bss_start__ lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __bss_end__ lld: warning:

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

Hello! Asterisk "CVS-HEAD-08/06/04-14:55:13" won't start on two of three different Gentoo machines. This is the output of gdb: ultra asterisk # gdb /usr/sbin/asterisk GNU gdb 6.0 Copyright 2003 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions.

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Hi, I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following, WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue

Hello all! My basic problem is that we haven't been able to get realtime to use ODBC to store configuration data. Here are the details: We've installed Asterisk on a CentOS machine as follows: 1. Downloaded, compiled, and installed FreeTDS 0.63 2. Downloaded, compiled, and installed unixODBC 2.2.11 3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons, and Zaptel from CVS