similar to: Extensions.conf for basic IVR?

Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? 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, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). => When a call comes in, I'd like an AGI

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 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 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.

So, people have recommended building a system from scratch, start with a CentOS base and installing asterisk and all of the other utilities. I've only used Trixbox for my business system. I'm wondering what surprises I'd run into. Right now, I know I'd need the OS, Asterisk, something like FreePBX, I have a x100p card so I'd need Zaptel, does that come with asterisk?

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

Hello I'm using a standard Asterisk install with default settings, and when I run "reload", I see that Asterisk fetches configuration information from a lot more sources than just my extensions.conf and sip.conf. For instance: -- Registered indication country 've' -- Registered indication country 'za' -- Setting default indication country to

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.

Can I do this? I have a x100p card on my PSTN line and I have an incoming context for these calls which uses the "s" extension. I'm wanting to set up a simple IVR and would like to be able to test the dialplan as I go. But having to dial-in on my PSTN line each time is going to cost me money. Can I connect to my zap_incoming context from my locally connected SIP phone? I'm

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 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 =>

hi, i'm trying to set a dovecot/pop + postfix couple with ldap authentification. and i'v got a lots of troubles. hers's my dovecot.conf protocols = pop3 login = imap login = pop3 first_valid_gid = 1000 last_valid_gid = 0 mail_extra_groups = mail default_mail_env = maildir:/var/mail/%1u/%u/Maildir mbox_locks = dotlock flock auth = default auth_mechanisms = plain auth_userdb = ldap

Hi first project with samba, first post on this mailing list and, actually, first time using a mailing list ever ;-) In summary ========================================== I try to install samba in a test VMs, everything seems fine but when windows client joins the domain it doesn't register in samba's internal dns in detail ========================================== My objective -

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.

My "application_helper.rb" become to large. I have basic functions that I use for all projects. Is it possible to define thoses functions inside an external "basic_helper.rb" and include it in "application_helper.rb" ?

I have hosts connected to interface INTERNAL with IPs that are on the public interface PUBLIC. Zones are "net" & "loc". I have proxy-ARP turned on for thoses hosts. The firewall is denying packets going to those proxied hosts, which is logical. What is the appropriate rule to allow it? In my mind, this is sort of "net" to "net", which

I cannot connect MuSe (using line in) to my icecast server. I get thoses messages repeting : Internal buffer inconsistency. flushbits <> ResvSizebit reservoir error: l3_side->main_data_begin: 1640 Resvoir size: 1792 resv drain (post) 5 resv drain (pre) 0 header and sideinfo: 120 data bits: 679 total bits: 804

Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy