similar to: Offhook tone in channel OSS/dsp

Displaying 20 results from an estimated 2000 matches similar to: "Offhook tone in channel OSS/dsp"

2005 Jul 27
0
Sending DTMF Tones Offhook
Greetings All! The Asterisk Call Manager works great. But I have one question for anyone who has used it. I cannot get the system to send some DTMF tones down the channel once the call has been made. Below is the script I am using to make the call, and start recording the channel. I am starting to make a system the will use asterisk to become an automatic random quality monitoring system
2005 Aug 18
0
help with waning on OSS/dsp, condition 16 and 17
Hi, I hope someone could help me, or at least give me a hint about this error. ERROR => Aug 18 10:48:51 WARNING[17120]: chan_oss.c:838 oss_indicate: Don't know how to display condition 16 on OSS/dsp -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Aug 18 10:48:53 WARNING[17120]: chan_oss.c:838 oss_indicate: Don't know how to
2004 Dec 06
0
TDM OnHook/OffHook
My TDM400P w/ 4 FXO cards seems to have trouble with onhook/offhook switching. It dials perfectly, but does not seem to be changing the onhook/offhook state appropriately. It changes sometimes, but it's not really reliable. For example: When I booted the machine, it started as onhook. It remained "onhook" through the entire first call (which was silent on both ends --
2005 Sep 23
1
FW: channel offhook state
> -----Original Message----- > From: Jacqueline Lee [mailto:jlee@isdomaininc.com] > Sent: Friday, September 23, 2005 11:46 AM > To: asterisk-users@lists.digium.com > Subject: channel offhook state > > > We are using a digium card (TDM400) with asterisk for our access to the > PSTN. Initially when the server starts, all the zap channels on the card > are in the
2004 Dec 07
2
TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be
2003 Aug 22
0
dtmf/audio before going offhook
Hello, Caller -> PRI -> SIP -> application. If application goes offhook right away, dtmf/audio works fine in both directions. If application, before going offhook (sending OK) plays a message and wants dtmf/voice from the caller, then caller can hear this message but his dtmf/voice don't reach application. Any way to configure it? Thank you. Alex Zarubin -------------- next
2004 Aug 03
1
Analog channel stays offhook
Hi, We are having a problem with asterisk detecting that an analog ext has been put down. This seems only to happen after a number of calls have been made. We have an FXO port (TDM400P with FXO module) connected to our PBX and are using this to test asterisk prior to rolling our for our small office. What happens is that we make a number of calls to this ext which 1st rings a phone (FXS)
2004 Dec 15
2
TDM400p FXO module always offhook
I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with what seems to be correct settings (according to digium and asterisk wiki). As soon as I plug in my POTS line into FXO mod the line goes into offhook state (whether I have power to the card or not). Should this happen? When I try to call * box all I get is busy signal. I've installed stable version, cvs version, change
2004 Dec 04
0
x100p offhook/onhook states
Hi, I'm having an interesting problem with my card. It seems to work fine, for the most part. When I first load the module and asterisk, it detects the line in the on-hook state. However, after the first phone call, zap show channel 1 lists it as being off-hook. During subsequent calls, the card is listed as being on-hook, and when it's not used -- off-hook. There are also some weird
2005 Jul 13
5
CONSOLE/dsp
I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for
2004 May 07
6
X100P keeping PSTN line Offhook
Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! ======================================== pbx1*CLI> zap
2008 Oct 20
1
Zaptel FXO offhook when connected to PSTN
I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*
2005 Aug 12
3
TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area
2006 Mar 27
2
How to disable event_log?
Hi, how can I disable event_log in order to reduce hard disk activity? I can't find any hints in conf files. Must I hack the source code or even use brutal methods like creating a dir called event_log in the log dir, in order to prevent asterisk from creating an event_log file? (Just chmod a-w event_log does not work, unfortunately.) Thanks for any hints! Roger.
2004 Aug 27
2
how to fetch a call?
Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's "einen Ruf heranholen". It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to
2005 Jan 03
3
oh323 context for peers
I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323
2010 Nov 14
0
freebsd oss sound dsp scheme
Hi I had have trouble to get sound working under freebsd 8.1 amd64 arch. So i decide to dig in wine code and I create a simple "proof of concept" patch to get work my sound card.? ********* SND STAT ********FreeBSD Audio Driver (newpcm: 64bit 2009061500/amd64) Installed devices: pcm0: <HDA Realtek ALC272 PCM #0 Analog> (play/rec) default <- This is my primary snd pcm1: <HDA
2005 Aug 15
1
permission denied when monitoring channel OSS/dsp
Hi! When I want to monitor the OSS/dsp channel through the Asterisk management interface, I get a "permission denied" error: Action: Monitor Monitor: OSS/dsp File: 1124096949 Mix: 1 Response: Error Message: Permission denied My permissions for /var/spool/asterisk look like this: pound:~# ls -la /var/spool/asterisk/ total 40 drwxr-xr-x 10 asterisk asterisk 4096 Aug 9 10:19 .
2011 Nov 13
0
how to get dev/dsp and oss modules back
Hello I have some older programs that require OSS and /dev/dsp I have tried the pulseaudio trick with padsp but some work and some don't. I have also edited the /etc/modprobe/dist-oss-conf and uncommented the line that says install snd-pcm /sbin/modprobe --ignore-install snd-pcm && /sbin/modprobe snd-pcm-oss && /sbin/modprobe snd-seq-device && /sbin/modprobe
2012 May 15
2
OSS DSP sound card input on CentOS 6.2?
Hello everyone, I'm streaming audio on CentOS 5.8 with no problem, even on a cheap sound card using DarkIce as the input tool. For the input under CentOS5, I use: device = /dev/dsp # OSS DSP soundcard device for the audio input But under CentOS 6.2, there is no such device. I see /dev/snd, and it has: controlC0 hwC0D2 midiC0D1 pcmC0D0p pcmC0D2p pcmC1D0p seq controlC1 hwC1D0