Displaying 8 results from an estimated 8 matches for "aelchuk".
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elchuk
2004 Aug 24
3
Hardware for PBX with 4 incoming/outgoing lines and 20 phones
Hi
I am interested in setting up an Asterisk PBX in my office with digium
hardware, and I just have a few questions in regards to what I would
need. It is my understanding that an FXO card is used to interface with
an incoming/outgoing phone line, and an FXS card is used for interfacing
with a phone within the system. Currently we have 4 incoming/outgoing
phone lines and would like to have
2004 Jun 30
1
Sound not working?
When I call into my system I have it set to play a bunch of different
sound files (I'm doing testing right now), but when it connects there is
just nothingness for about the time the sound file should take to play.
A soundcard is installed on my system and working properly. Last night
I would sometimes get "Sound: Record Overrun" messages when I would
close asterisk down or
2004 Jul 01
0
Sound: Record Overrun
Hi,
When I dial into asterisk I set it up in extensions.conf so it will play
some messages, but when I dial in asterisk picks up but I hear no
sound. There is moments of silence where the audio should be playing
but I get nothing. I checked /var/log/messages to see what was wrong
and I got the following error:
Jun 29 20:46:33 eclipse kernel: Sound: Recording overrun
Does this mean
2004 Jul 06
0
Sound card troubles with asterisk resulting in no sound
Hi,
I have a redhat 8 linux box with a an X100P and a soundblaster sound
card. When I start asterisk I get the following message:
WARNING[65544]: chan_oss.c:238 sound_thread: Read error on sound device:
Resource temporarily unavailable
And then if I try to call in none of the audio files I have set to play
in extensions.conf actually play. When I'm outside of asterisk and try
playing a
2005 May 13
0
Dropped Calls between Sip and Zaptel
Hi,
I am having trouble with dropped calls in Asterisk. I've done a bunch
of searching but all I could find was setting busydetect and
callprogress to yes in zapata.conf to help combat the problem, but I did
this to no avail. The following is the output from
/var/log/asterisk/full at the time the call was dropped on me.
May 13 08:37:13 DEBUG[5379]: Stopping retransmission on
2005 May 17
0
Dropped calls with TDM400P - 4 FXO
Hey,
I've done some searching for this and never really found a concrete
answer. Is there a specific reason or solution why just in the middle
of a call Asterisk will drop it and I'll get dial tone again? Anyways,
this is the output from /var/log/asterisk/full at the time of disconnection:
May 13 08:37:13 DEBUG[5379]: Stopping retransmission on
2004 Jun 28
4
Dial Command
I'm trying to use the dial command to initiate a call to number 9661443
with an X100P card set up on channel 1 with the following in my
extensions.conf:
exten => 1,1,Dial(Zap/1/9661443,15)
Then when that command executes in the asterisk daemon I get the following:
app_dial.c:688 dial_exec Unable to create channel of type 'Zap'
Can anyone tell me what might be wrong?
2004 Jun 29
4
Getting Asterisk to automatically dialout
Hi,
I'm trying to get asterisk to auto-dail out. I created a *.call file
with the the top of it being "Channel: Zap/1/2609944", which should have
connected to Zap channel 1 and dial out to 2609944, but It did not do
so, asterisk would say a call was completed to Zap/1/2609944 but I never
heard that phone ring. So I tried just putting "Channel: Zap/1" at the
top of