similar to: User unable to use DTMFs?

Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you.

Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then

Hello I've been googling for this message, but can't find why Asterisk sends a warning. The configuration files look similar to http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample It's a TDM card with just one FXO module on it, and I connected an RJ11 cable to it from the wall plug: # cat /etc/zaptel.conf fxsks=1 loadzone=fr defaultzone=fr # ztcfg -vv

Hello When using LookupCIDName, Asterisk 1.4 says that it's deprecated, and we should use "${DB(cidname/${CALLERID(num)})}" instead, but I don't know how to use it: ;DEPRECATED exten => s,1,LookupCIDName ;ERROR exten => s,1,${DB(cidname/${CALLERID(num)})} I guess I should use this as a parameter to a function, but which one? Thank you.

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

Hello, I'm looking for a dirt cheap solution for SOHO use to handle at most a couple of POTS lines, and I notice that X10?P cards go for $15 on eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma. I have a couple of questions about those cheap FXO cards: 1. Are they all glorified softmodems, ie. none has an on-board CPU or DSP and outsources all processing to the

Hello I've tried a bunch of things, but still get errors/warnings when using the IF() function: ============== TEST #1 exten => h,n,Set(WAV_FILE=${IF($[ ${STAT(e,/tmp/${CALLTIME}.wav)} ]?${CALLTIME}.wav)}) [Nov 26 21:52:34] WARNING[5074]: func_logic.c:107 acf_if: Syntax IF(<expr>?[<true>][:<false>]) ============== TEST #2 exten =>

Hello, I'm trying to fill CID data into the astdb using AsteriskWin32's asterisk.exe, to no avail: The batch file stops after the first line, and just waits: ---------------------------------------- rem c:\cygroot\mystuff>import.bat rem rem c:\cygroot\mystuff>C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 "My cellphone"' rem rem Asterisk module

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

Hello I've never built an IVR before, so I was wondering if someone could share some code from their extensions.conf that would perform some of thoses steps: 1. When a call comes in from the TDM FXO port, answer 2. If no CID, play message "No CID available. Please type the number where you wish to be called back". Loop until OK or remote party hung up 3. When CID is available,

Hi Most of our customers have generic names like "Hospital", so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as "Reading Hospital" so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you.

Hello This PC had been running a Ports-compiled Asterisk 1.4.16.x succesfully for almost three months, but this morning, although Asterisk itself seemed fined, the Zaptel interface stopped taking calls. Stopping/restarting Zaptel using /usr/local/etc/rc.d/zaptel stop-start didn't let things recover. Since I didn't know better, I had to reboot the host to get things working. I used this

Hi, I've just installed DAHDI at two PBXs as follows: *PBX-1 PBX-2* FXO ------------- FXS When I try to send calls from PBX-1 to PBX-2 I just receive the message: "Starting simple switch on 'DAHDI/1-1" It seems like if PBX-1 couldn't send DTMFs, but when I set immediate=yes at chan_dahdi.conf at PBX-2 dialplan is executed at PBX-2 but nothing is heard at

Hi SIP is such a pain to use when NAT is involved that I'm willing to buy an IAX hardphone for someone who works remotely over the Net and needs to get calls from our Asterisk server, itself behind a NAT. Are there good, affordable IAX phones you would recommend? Thank you.

Hello I've been googling for a couple of days now, but still can't figure out what to put in zapata.conf to get it to report CID. Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202 as CID FSK Standard: http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium

Hello Now that I have my first IVR up and running, I'd like to have Asterisk create tickets in a bug tracker every time a call comes in. It's a nice way to know who's calling and why, before following up on the cause for the call. There are tons of bugtracking apps out there. Do you know of some that I should look at? Ideally, the interface shouldn't be much busier than JoS

Hello I'm trying to build the Zaptel ports on FreeBSD, but it fails: ========== # cd /usr/ports/misc/zaptel/ # make install ===> Building for zaptel-1.4.6_3 make -C zaptel all Warning: Object directory not changed from original /usr/ports/misc/zaptel/work/zaptel-bsd-1.4.6/zaptel cc -O2 -fno-strict-aliasing -pipe -Wall -D_KERNEL -DKLD_MODULE -std=c99 -nostdinc -I. -I@ -I@/contrib/altq

Hello Even the cheapest GSM gateway I found for a simple SOHO server costs over $300. If possible, I'd rather have a local solution than use a remote VoIP provider to have Asterisk make outgoing calls to the GSM network. So... I was wondering if there are some entry-level cellphones that can somehow be hooked up to a PC running Asterisk and used as an el-cheapo GSM gateway? Thank you.

Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively.