similar to: Voice mail not working with Asteriks 1.2.5

Hi to all, I have been using Wildcard TDM400P with four fxs modules and it was working fine now I have added another Wildcard TDM400P with four fxs modules . So there are total 8 ports for 8 hard phones. I have modified following configurations In /etc/zaptel.conf loadzone=us defaultzone=us fxoks=1-8 in /etc/asterisk/zapata.conf context=headoffice signalling = fxo_ks

Hi, All of you people, I just want help on issue regarding auto dialing in asterisk. I have implemented asterisk server using TDM400P with four FXO Modules, as well as I am using analog phones. That's works fine, can any one of you will guide me, how can I implement basic auto dialing functionality in which I will be storing list of phones number either in GUI interface or will feeding in

Hi all of you. I am trying to configure voice mail in asterisk and i am facing problems. I have found following warning message in /var/log/asterisk/messages -------------- No application 'Voicemail' for extension (macro-mainmenu, s, 5) I have configured voice mail accordingly in extention.conf [headoffice] -- ------------ ------------- exten => _63,1,Macro(mainmenu)

Hi to all, I and using asterisk with following setup. 1. TDM400p card with four FXS modules, so there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 I want your guidance for the following issue. To ensure that CLI or called number is attached to a database which automatically updates records of the

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

Hi to all, I and using asterisk with following setup. 1. TDM400p card with four FXS modules, so there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 I want your guidance for the following issue. with help of agi scripts i am able to insert caller id number in database of mysql now i want to pop it up via

Hi to all, I and using asterisk with following 1. TDM400p card with four FXS modules, So there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 But when some dials form his number (suppose abc) to my number (suppose xxxx) I get abc number on my analog phone, but now I have purchased more than one numbers

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:

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

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 updated asterisk today via svn no i can'T start asterisk i get core dumps. i have to comment some modules then i can start: noload => format_au.so noload => format_mp3.so noload => format_pcm_alaw.so.so noload => format_pcm_alaw.so compiling was fine just some warnings somebody has any idea? -------------- next part -------------- An HTML attachment was scrubbed...

I have an ext3 filesystem that I want to mount without loading the journal. I tried the "noload" option with both 2.4.9 and 2.4.18 series kernels and get the errors listed below. [root@host]# mount -t ext3 -o noload /dev/sdf1 /mnt mount: wrong fs type, bad option, bad superblock on /dev/sdf1, or too many mounted file systems /var/log/messages: Apr 18 13:30:23 host kernel: ext3:

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

I am newest to this group and would appreciate your help! Is it possible to use quicknet phone jack with asteriks@home ver 0.6? Little has been mentioned about use of quicknet products' adaptability with asteriks@home I do have a couple of old jacks to startup right away. Your guide is most welcome. Thanks, Mike __________________________________ Celebrate Yahoo!'s 10th

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

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

We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on "attended transfers". What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to