similar to: Re: Budgetone and MWI

asterisk-users-request@lists.digium.com is believed to have said: >Budgetone and MWI > >The message button can be programmed to dial an extension that checks >voicemail >exten => 160,1,Voicemailmain(${CALLERIDNUM}) > Thanks, this is what I was thinking about. Still, how do you get the BT to dial 160? In my Asterisk setting I have the same mailbox numbers reused for the

I tried to use message waiting indicator, by "Subscribe for MWI" in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald

asterisk-users-request@lists.digium.com is believed to have said: >Ronald, it's the context listed in voicemail.conf (I got caught on this >as well) > >I really wish Asterisk was better documented; it's bullshit the way it >stands at the moment. > > >Cheers, >Dean > Dean, so if I have two contexts defined in voicemail.conf, like: [general] [local]

Hi all, sorry bothering again. I am still stuck in compiling asterisk. Learning (or trying to) from the first problem and first hint, when I got this error: gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o

Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out

Dear all, I have tried to compile * 1.0.6 (downloaded from the digium site, in the right sequence - zaptel, libpri, asterisk) on two different machines running SUSE 9.2. The problem comes during some preliminary checks: checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in

Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). ***** A few details on the box: cat /etc/redhat-release CentOS release 5.10 (Final) arch i686 uname -a Linux hermes 2.6.18-371.4.1.el5 #1 SMP Thu Jan 30 06:09:24 EST 2014 i686 athlon i386 GNU/Linux asterisk -r Asterisk 1.6.2.20, Copyright (C) 1999 - 2010 Digium,

asterisk-users-request@lists.digium.com is believed to have said: >Hi, > >Sorry for being very late on this thread but i am trying to make a >decision on which one to go for. Options are > >1. Dock n Talk offered by Voxilla (USD139) >2. GSM Gateway by CyberTelecom (GBP60) > >I'm having a TDM400P with 1 FXO & FXS. > >I'm interested in implementing DISA

asterisk-users-request@lists.digium.com is believed to have said: >Bluetooth enabled phones to talk to Bluetooth headsets; I guess there's a >protocol for the phone to talk to any Bluetooth headset, no matter who made >it. This protocol would have to include something to allow voice to pass >from the phone to the headset and vice versa. It might also include >something for

I have found that the NTP server is not contacted when the phone (Budgetone 100) comes back from a power down. I must reboot the phone without powering down to get the phone to contact the NTP server for the time. It doesn't matter whether I reboot from the phone's web site or using the menu reset function: either works. I have only tested this with a private NTP server. This may

I know that this subject has been treated in the past! As a matter of fact reading some old messages about compiling zaptel I made a couple of tests after the first compiling failure to understand why I can't compile on a specific machine, but I do not know how to handle the results. The machine has SUSE 9.3, and an updated kernel (2.6.11.4-21.7-default; as shown below). YAST (the graphical

Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb

Hi all, I have noticed something I can't solve regarding Asterisk (latest 1.6.0.x). My server is set at the GMT+2 timezone. The clock is ok (I can get the correct time at the terminal). But today I got a call at a time where Asterisk should have gone 'off business hours'. All log times are wrong by exactly 2 hours. As if Asterisk would just sit on GMT, ignoring the GMT+2

Of course I am not a kernel expert, so .. please be patient. I am investigating on my zaptel/zapata problem. As the main error message asterisk quits on mentions <'/dev/zap/channel': No such file or directory> I went peeking over there. [Asterisk Verbose Error Mar 13 20:43:35 WARNING[5779]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or

Hi I was wondering if there are any special settings that I need to be able to transfer calls. Whenever I press the 'recall' button, I just here a click, and no ring-tone to transfer. in my debug log I get this : -------------------------- Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1 (index 0) Feb 24

Dear list, I tried to install the CAPI support RPM from the SUSE 9.2 disks. YaST tells me it is happy with the process. The Asterisk release I am using is the one that comes packaged in RPM format, also included in the distribution. Still starting asterisk with the usual asterisk -vvvc I see that something goes wrong. [app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242

asterisk-users-request@lists.digium.com is believed to have said: >I am using slackware 10.1 (kernel 2.4.29) and I am getting the following >when I issue gcc -v Dimitris, while I never compiled chan_capi I thought you would need a 2.6 kernel to use it. HTH Aldo

asterisk-users-request@lists.digium.com is believed to have said: > >and watch linus himself rant about how this is incorrect to do (yet all >the distros do it) :P > Well, this is reassuring for a newbie like me. Even the pros (as anybody building a distro ought to be, and most of the times, really is) can do obvious errors... Aldo

asterisk-users-request@lists.digium.com is believed to have said: > >The active cards do the ISDN protocol stuff on board, so the host CPU/driver >does not need to do that -> better performance, less interrupts. >The AVM cards do not have such DSPs on board, so no echo-cancel. >But the Eicon DIVA Server cards do. They do analog Fax/Modem, echo-cancel, >DTMF-detection, voice

Hi, I am seeking help with the installation of a Soundpoint 650 desk phone. Although I have some experience (and a good one! no single issue so far, besides the problem I am trying to solve...) installing a few SP 320/330 units, I am having several issues with my first SP 650. Polycom SP 650 Data: ? P/N: 3150-11530-212 ? SD Sound ? FW: 2.1.2.0078 ? Assembly: