similar to: SayUnixTime

Hi Gang, Hopefully somebody out there has a "doh" for this one. My dialplan announces the date and time using SayUnixTime. When I run this: exten => 36225,1,Set(ABA=999999999) exten => 36225,n,Background(telbank/${ABA}/${CHANNEL(language)}/thetimeis) exten => 36225,n,sayunixtime(,,Abe 'digits/at' IMP) I get this CLI output -- Executing

I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B (603, 604). I have two lines on the TDM22B. I cannot figure out some of the problems: 1. 601 dials via ZAP/3-1 to local phone number at PSTN: ringing pickup on PSTN (empty) still ringing in the phone set 601 2. call from PSTN back: 601 picks up ... everything works !!! No caller id shows up 3. For testing I have only one

Can anyone tell me what I missed? I'm trying to setup a simple extension (400) that reports the time when it is dialed. I searched the threads and it seems like this should work... Here's what's in my extensions.conf: exten => 400,1,Answer() exten => 400,n,Wait,1 exten => 400,n,SayUnixTime(,EST5EDT,) exten => 400,n,Playback(tt-weasels) [BTW, tt-weasels is hillarious!

I noticed that the output of "rsync -n" is sometimes truncated. Even when you repeat the same command, the output may be different. I'm using rsync-2.5.5 on a Debian woody system with ext3 file systems. One scenario to reproduce this behavior is someting like this: # rsync -aR /usr/share/zoneinfo /tmp # rsync -n -avRH localhost:/usr/share/zoneinfo /tmp receiving file list ... done

I'd like to announce the time when people call and hit my voice-menu context, but the complaint is that the time announcement isn't interruptable. Is there any way to make SayUnixTime interruptable? -- PhoneBoy

Hi list users I?m trying to do an incluye statement from the Database In my dialplan I have different contexts that defines common services for example ---------------------------------------------------------------------------- ----------------------- [basic_services] exten => 100,1,VoicemailMain() exten => 600,1,Playback(demo-echotest) exten => 600,2,Echo() exten

I'm trying to get SayUnixTime to parse the date correctly and say 'May Twenty Fifth', however it refuses to do so and keeps using the phrase 'May Twenty Five'. I've read all the syntax, tried using both 'd' and 'e', and in the doc it says: d or enumeric day of monthfirst, second, ..., thirty-first However, it doesn't read my h-#.gsm files...

I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): 1. Sip Config Mode: Proxy Primary Proxy IP Address: *.131 Primary Proxy port: 5060 Line1 Number: 1002 2. Security Config Line1 Account: 1002 Line1 Password: ****** 3. Line Configuration Line1: Type=FXO, Hunting Group=2, Hot Line =

before I accuse somebody to "overbill" I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers: 0.7269 or 0.2929 ??? bye Ronald Wiplinger

##### BEGIN CODE ###### #!/usr/bin/perl ###### # # --start, -s = The date you would like to start generating regressors #--end, -e = When to stop generating holiday regressros # --scope, -c = D, W for Daily or Weekly respectively (e.g. Does this week have a particular holiday) # --file, -f = Ummm where to write the output silly! # # **NOTE** The EOM holiday is "End of Month" for

Here is the correct version. The old version is the redirect only version of the script. ### BEGIN SCRIPT #### #!/usr/bin/perl ###### # --start, -s = The date you would like to start generating regressors #--end, -e = When to stop generating holiday regressros # --scope, -c = D, W for Daily or Weekly respectively (e.g. Does this week have a particular holiday) # --file, -f = Ummm where to write

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

James, thanks for the explanations. I misread the notes. As an exercise, I'm trying to convert an existing project that currently uses Scriptindex and Omega to direct Xapian API calls. I did a (I think) complete dump of a document with delve -r 565 -d database and I see things like subject='A typical subject' with a corresponding set of terms like Sa Stypical Ssubject Which

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing

Below is my extensions.conf for the fax part [incoming_28345474] ; ;******************************************************************** ; BEGIN - Inbound call handlers ;******************************************************************** ; exten => 8862100,1,NoOp(${CALLERID(num)}) exten => 8862100,2,Background(if-u-know-ext-dial) exten =>

I tried to add the databases from the localhost to the database server and changed the every /etc/asterisk/*.conf from host=localhost to host=192.168.10.10 (my dababase server) When I restart asterisk, I do not get any errors, but after a phone call I see: Jun 7 18:11:56 ERROR[7877]: cdr_addon_mysql.c:400 my_load_module: Failed to connect to mysql database cdr on 192.168.10.10 Or if I try

Skype is very succesfsfull and get more and more users, ... we can ignore them, accept them or do something,... My suggestion is that we try to do something, ... If we would peer to each other, than we get soon also a great amount of users together, and than our service becomes more valuable, ... Let's discuss advantages and disadvantages! bye Ronald -- Ronald Wiplinger (CEO of

Can anybody spot the problem? H.323 via GK calls Asterisk box and should be connected directly to the extension 6002 (an voice sip phone) The caller hears only ringing. The called party hears the ring. Called party picks up, caller hears still the ring tone. Called party hears nothing. Called party hangs up, caller hears the busy tone. extensions.conf: exten => 88670333333,1,Wait(1) exten

You'll need to create a trunk between the two systems Then configure your out bound routing to use that trunk Eg if you use 1 as the prefix for the trunk BOX A Ext200 will call box B EXT201 by dialing 1201 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ronald Wiplinger Sent: Sunday, 19 June 2005

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald