similar to: Soundcard necessary on an asterisk server toget output of playback()?? -> Next step

Displaying 20 results from an estimated 1000 matches similar to: "Soundcard necessary on an asterisk server toget output of playback()?? -> Next step"

2007 Dec 04
1
Soundcard necessary on an asterisk server toget output of playback()??
Hi, >However, I believe that zaptel >= 1.4.6 or zaptel 1.2 >= 1.2.21 should >support hires timers for timing on kernel >= 2.6.22 . > >What version of Zaptel do you use? > I was using version 1.4.5.1 I just downloaded and installed version 1.4.7, configure/make/make install finished without an error, but when is used modprobe ztdummy the system said: FATAL: Error
2008 Jan 13
0
Soundcard necessary on an asterisk server to get output of playback()?? -> Next step
Tzafrir Cohen wrote: > > The agent picks up the phone but neither the agent nor the caller > > > here anything. >So please provide a more complte trace and a the relevant partt of your >dialplan. > Here is the relevant part of the dialplan: [local] exten => 98,1,Dial(SIP/sguenther,20,tr) exten => 98,2,VoiceMail(98|u) exten => 98,3,hangup exten =>
2007 Dec 03
4
Soundcard necessary on an asterisk server to get output of playback()??
Hi, I' still fighting the problem, that I can talk from one SIP phone to another, but I can't hear the output of the playback or similar applications: exten => 202,1,ANSWER() exten => 202,2,PLAYBACK(tt-monkeys) exten => 202,3,HANGUP() When I dial 202, asterisk show the following on the cli: -- Executing [202 at local:1]
2007 Nov 12
0
No sound from playback and voicemail (Atis Lezdins)
Hello, >> > I can talk to other SIP phones and via ISDN to the outside, but I >> >don't hear playbacks or the voicemail messages. >> > Asterisk show in the cli, that the corresponding files are played, >> >but I hear nothing at all. >> > >> > Here is as simple example: >> > >> > [monkeys] >> > exten =>
2007 Nov 12
0
No sound from playback and voicemail (Carlos Chavez)
>On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther wrote: >> > Hello, >> > >> > I have a strange situation: >> > >> > I can talk to other SIP phones and via ISDN to the outside, but I >don't hear >> > playbacks or the voicemail messages. >> > Asterisk show in the cli, that the corresponding files are played,
2007 Nov 12
3
No sound from playback and voicemail
Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] ??? exten => 99,1,ANSWER() ??? exten => 99,2,PLAYBACK(tt-monkeys) ??? exten => 99,3,HANGUP() The phone
2009 Jul 10
3
strange strsplit gsub problem 0 is this a bug or a string length limitation?
I was working with the rmetrics portfolioBacktesting function and dug into the code to try to find why my formula with 113 items, i.e. A1 thru A113, was being truncated and I only get 85 items, not 113. Is it due to a string length limitation in R or is it a bug in the strsplit or gsub functions, or in my string? I'd very much appreciate any suggestions ============Input script:
2006 Feb 21
0
Trusted domains within a large enterprise
Hi Folks I need some advice on whether what I am doing is correct, initially from a logical perspective. My company (E.ON - large utility) has a large ADS system. We are retiring NT4 domains and I have been asked to transfer the SAMBA domain log-ins into ADS. I am initially testing my work on Linux RHEL 4, running SAMBA 3.10. The ADS system consists of a realm/forest PG.EON.NET (old
2023 Jan 11
0
New CRAN package announcement: azlogr
I am excited to announce that `azlogr` is published on CRAN! This package?enables logging in 'R' by extending the functionality of 'logger' package. There is an option to add additional custom meta-data while logging, which can sometimes be helpful.?Logging messages are displayed on console and optionally they are sent to 'Azure Log Analytics' workspace in real-time. The
2023 Jan 11
0
New CRAN package announcement: azlogr
I am excited to announce that `azlogr` is published on CRAN! This package?enables logging in 'R' by extending the functionality of 'logger' package. There is an option to add additional custom meta-data while logging, which can sometimes be helpful.?Logging messages are displayed on console and optionally they are sent to 'Azure Log Analytics' workspace in real-time. The
2011 Jul 29
2
problems with image.plot()
Hi all, I used image.plot() to create a heat map of a matrix: as.matrix(read.table("Matrix.txt", sep="\t"))->x HeatBrk<-seq(5,25,2.5) MyCol= gray((7:0)/7) library(fields) image.plot(x, col=MyCol, breaks=HeatBrk, legend.shrink=0.3) dev.copy(device=pdf, file="HEAT4!.pdf", height=8, width=8) dev.off() There are a few things that I would like to do that I
2004 Aug 30
7
Polycom SoundPoint IP 300 Configuration
I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up Anyone willing to point out a good asterisk
2006 Apr 01
8
ROR on solaris 10 gem installation error
On my Sun Enterprise 450, I installed ruby-1.8.4 from sunfreeware.com. That installed like a dream. I then downloaded rubygems-0.8.11. Untarred into /usr/local/src and tried to do a /usr/local/bin/ruby install.rb this is what followed and can someone help me figure out what to do? bash-3.00# /usr/local/bin/ruby setup.rb ---> bin <--- bin ---> lib ---> lib/rubygems <---
2008 Feb 15
0
Question about DIALSTATUS NOANSWER
Hi, according to the wiki the value NOANSWER for the channel variable DIALSTATUS means: No answer. The dial command reached its number, the number rang for too long, then the dial timed out. In out dialplan we grap all these events with exten => s-NOANSWER,1,Playback(sometext) exten => s-NOANSWER,2,WAIT(1) exten => s-NOANSWER,3,Hangup() The dial commands for internal and external
2010 Mar 21
0
dahdi_monitor doesn't show data on RX & TX: broken card or cable?
Hi, on one of our clients asterisk server we have the problem that you hear nothing on external calls. Here are the details abount the system: Asterisk 1.6.0.22 DIGIUM Wildcard B410 quad-BRI card (rev 01) dahdi-linux-complete-2.2.0+2.2.0 I have setup the following test extension: exten => 9216992,1,ANSWER() exten => 9216992,2,WAIT(2) exten =>
2007 Dec 18
0
Doorbell Siedle DCA 612 and Asterisk?
Hi, has anyone already set up a configuration between the doorbell Siedle DCA 612 and an Asterisk Server? I have used a Grandstream HT 286 to connect the doorbell and the asterisk. When I press the button, the phone ring and when I pick up the call I hear a beeping. At the door I hear nothing. According to the wiki, this doorbell should work with Asterisk, but I haven't found a dialplan
2008 Feb 05
1
Mistake in the wiki's description of cmd Pickup() ?
Hi, according to the description of Pickup() on page http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup I can use this command to pickup a call at a certain extensions. When I try this with e.g. exten => *8200,1,Pickup(200) Asterisk tells me that the highest value for the Pickup command is 63. Wenn I enter the number of a callgroup instead of an extension, I can pickup the call.
2008 Feb 04
0
Problem picking up a call with PickUpChan or PickUp [SOLVED]
Paul Madley wrote > >Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 >release, and therefore I don't think any config changes will fix it. >We've been told to roll back to our previous 1.4.13 installation. It >also seems to manifest itself in "ghost ringing" as I've called it; >place a call to a SIP extension, then put down the
2007 Jul 31
0
Number disappears when picking up a call
Hi, when I pick up a call with *8, the number of the caller isn't show on the phone that picked up the call. Is there a way/chance to keep or transfer the number of the caller? We are currently using Asterisk version 1.4.1. Thanks for any hints, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133
2007 Aug 08
3
Siemens Openstage & Asterisk ?
Hi, is anyone on the list using the Siemens Openstage phones together with asterisk? If yes, is it possible to use the programmable keys of these phones together with Asterisk? Thanks for any hints, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93