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...would I need to edit sip.conf as well or some other configs or is that even possible? Thank you again. -- Andrew Elchuk Technical Associate Cronus Technologies 248 - 111 Research Drive Saskatoon, SK S7N 2X8 Tel: (306) 652-5798 ext. 112 Fax: (306) 652-5799 Toll Free: 1-877-655-5798 http://www.cronustech.com

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

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

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

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

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

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?

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