similar to: Can I enter an extension to dial while voicemail is playing?

The options are not seperated by commas. exten => s,1,Dial(SIP/50,23,r,d) should be exten => s,1,Dial(SIP/50,23,rd) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Klimek Sent: Thursday, June 22, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Hi, I'm running asterisk 1.4.22 on a debian server. I have php5 installed and it works correctly command line. When trying to run a php script via AGI, I get messages such as: GI Tx >> I> AGI Rx << #!/usr/bin/php5 -q AGI Tx >> 510 Invalid or unknown command The scripts are completely executable and owned by asterisk -rwxr-xr-x 1 asterisk asterisk Googling is not helping

Hi! I am currently working on setting up an Asterisk system, and I was wondering if anyone has worked on an alternate mp3 player to mpg123. We have a library of MP3 files that we would like people to be able to select and play over the phone -- and this will require pause & resume, as well as fast forward / reverse (jump forward / jump back). It doesn't seem like mpg123 can do this. Is

We've been running an Asterisk-based phone system here in our office for a year and a half, and it's pretty much been running smoothly. One employee who works out of the office has a problem that she can't make outgoing calls on a temporary basis every so often (a few times a day). No one else has this problem, her settings are fine, and she regains the capability spontaneously with no

Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server.

Hey, I've got a data set (e.g. named Data) which contains a lot of variables, for example: s1, s2, ..., s50 My first question is: It is possible to do this: Data$s1 But is it also possible to do something like this: Data$s1:s50 (I've tried a lot of versions of those without a result) My second question: I want to do a stepwise logistic regression. For this purpose I use the following

I am using Trixbox 1.0(Asterisk 1.2.7.1)at a customer site. They whishes to change the default Voicemail behavior. Standard behavior No answer/Busy -> send to Voicemail Requested behavior No answer/Busy -> message that if you press 9 you will instead be cent to reception -> send to Voicemail or Reception if 9 pressed. I want this to always happen when Voicemail is invoked. How

Yesterday we moved one of our servers to a new IP. We updated DNS and various adapters configured to register to that server registered to the new IP correctly. All seemed to be well. This evening I discovered that with one exception, all of the adapters are getting a SIP/2.0 401 Unauthorized message back from asterisk. The exception is an Innomedia adapter -- Linksys PAP2's and (I

Hi, I am trying to read the "credit.lisp" file of the Japanese credit database in UCI repository, but it is in lisp format which I do not know how to read. I have not found how to do that in the foreign library http://archive.ics.uci.edu/ml/datasets/Japanese+Credit+Screening <http://archive.ics.uci.edu/ml/datasets/Japanese+Credit+Screening> Could anyone help me? Best

Hi, For a few weeks now, our asterisk server has been experiencing something very odd. From time to time, voicemail.conf would go blank. We finally tracked it down to happening when someone attempts to change their password. It seems the file is touched, but not written to, and we're left with a blank voicemail file. Permissions seem to be fine: -rw-rw-r-- 1 asterisk asterisk 12707

I'm trying to setup Asterisk on my Linksys WRT54G router and it appears to startup successfully (no errors) and it says it is listening on 0.0.0.0 port 5060, but I am unable to connect to it. I've tried "telnet localhost 5060" but it just says connection refused. I've also tried connecting from another machine on my network (eg. telnet 192.168.0.1 5060) but it also says

One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine !!!! What indicates that there is no fault on his Internet connection!!! He is using his

Hi, Pretty sure I'm missing something simple, but I've seen references to this feature but not found documentation for it: I have a queue set up so that many people are contacted (ringall) when a call comes in. I would like the answering party to confirm that he is a human being rather than cellphone voicemaill by pressing a digit. This is somewhat similar to the 2nd macro example

On Debian the Asterisk Makefile does /usr/sbin/update-rc.d asterisk start 10 2 3 4 5 . stop 91 2 3 4 5 .; which results in a /etc/rc2.d/S10asterisk being written. I think S10 is too early. bind9 : S15 mysql : S19 zaptel: S20 ntp : S23 What bothers me most is that mysql is not up when asterisk starts. That's a bad thing if there are #execs in your config files and if the scripts rely on

I have some data 'myData' in wide form (attached at the end), and would like to convert it to long form. I wish to have five variables in the result: 1) Subj: factor 2) Group: between-subjects factor (2 levels: s / w) 3) Reference: within-subject factor (2 levels: Me / She) 4) F: within-subject factor (2 levels: F1 / F2) 5) J: within-subject factor (2 levels: J1 / J2) As this is the

Dear all, I have data.frame object in R. I want to export it in tab-delimited file with several lines of header initiated with comment sign (#). I do not know how to do that in R. Could you please give helps on this problem? Thanks in advance. Best, Jian-Feng, ################################################################## The lines I want to write in the header lines look like, with words

Ok, I've done some more research and I don't think I want an FXO box... What I'd like to do is use BroadVoice (with their BYOD plan) and then run Asterisk on my WRT54G router. I'd also like to use my regular home phones without having to use a special "SIP" phone... (eg. I like my Vtech normal cordless phones) What do I need to buy to get this working? It sounds like I

Asterisk seems to register just fine with BroadVoice (asterisk -r, and then sip show registry shows sip.broadvoice.com is "Registered") ...but when I try to call my broadvoice number (from a cell phone), it rings one single time and then says "The party you are trying to reach is not available to take your call." This doesn't seem to be an Asterisk message but seems to be

Incoming calls from my Sipura 3000 don't seem to be correctly routing to Asterisk (or something?) Here is my Asterisk configuration for my incoming PSTN line: Code: [1000] type=friend host=dynamic context=incoming secret=6769 dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very Inside of extensions.conf, I have this: Code: [incoming] exten => s,1,Answer( ) exten =>

Hello, > I am setting-up a system to place outgoing calls for a certain > number of minutes (as allowed per the customer's account). I would > like to "break into" the long distance channel to announce "1 minute > left", etc. What asterisk command can I use to do this? > > Thank you in advance for your help. > > Chaya Rosenberg >