similar to: Voice Message Matching?

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

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

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: >I don't mean to be rude to everyone who responded to this question, but >I think that everyone is answering the wrong question. The point is that >the message waiting indicator doesn't light up, at all, ever. All that >happens when messages are waiting is that the display blinks and the >phone gives a

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

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

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

Hi, Here's the deal. When someone leaves me a voicemail message I want Asterisk to call me on my cellphone by dialing my cellphone number and tell me I have a message. Is this possible? Can anyone cite examples? Most commercial voicemail systems produced in the last 10 years can do this. Any help would be much appreciated. Regards, Mike -------------- next part -------------- An HTML

Hi folks, my goal is to access voicemail (there were some posts about this) but not by dialing numbers. As asterisk sends voicemails in e-mail, it's cheaper for us to read e-mails on our cell phone (3g, gprs), and the message is attached there. i've looking around in voicemail.conf and found: [general] ; Default formats for writing Voicemail format = wav49|gsm|wav my phone

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: >I have Asterisk set up at 2 offices, connected via an IAX trunk. My delema >is one person is always moving between offices. I have the dial plan set up >to ring phones at both offices but his voicemail box is at office A. His >phone at office A has the message indicator, however, he wants to also have >the message

I have a pretty standard setup with Asterisk acting as a PABX for a bunch of SIP handsets (in this case, SwissVoice IP10S). My users are complaining that when they forward their phones to their cellphones on unavailable (i.e. forward when no-answer), their cellphone only rings once or twice, and then Asterisk sends the call through to Voicemail. I'm using the standard extension Macro

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:

I have a Motorola Razr successfully connected to asterisk using a bluetooth dongle and chan_bluetooth. Here's some issues I've run across: - You have to ignore the instructions in bluetooth.conf, saying to run "sdptool search --bdaddr xx:xx:xx:xx:xx:xx 0x111F" to determine the correct channel to use for your phone. My phone reported Channel 7, but will not work with anything

Hello list, I want to listen to voicemails on my * box from a phone that is not local to my pbx. I.e., from my cellphone or my PSTN work line etc. I'm aware that I can forward VM to email or use a web interface but that is not always practical. Other than doing an IVR type arrangement or a phone number dedicated to VM access is there a way to do this? On my old POTS line I used to be

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

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