Posted: Tue Apr 04, 2000 11:13 am Post subject: [Asterisk] Problem
Hi, Everybody.
I tried to work with asterisk and have got some problem. Please help me to
fix it.
1. When i hangup the phone. Sometimes I got this message:
WARNING: File chan_phone.c, Line 673 (do_monitor): Dial tone write error
after that the sound tone is gone.
2. I change phone.conf to used => format=slinear and format=g723.1
but I didn't hear anything from voicemail message.
Posted: Wed Apr 12, 2000 5:29 am Post subject: [Asterisk] Problem
On Mon, 10 Apr 2000, you wrote:
My sound card is "Sound blaster vibra 128". I'm very sure that it's full
duplex and i tried to work with Quicknet Internet phonejack too. I've 2 phone
jack and 1 sound card. I'm not sure have something wrong with my linux
redhat that i used because i can't recording with this sound card.
Another case i call from phone1 -> phone0 and waiting until timeout but I
didn't hear anything on phone1. Can you suggession me to do something...?
== Accepting call on 'Phone/phone1'
-- Executing Dial("Phone/phone1", "Phone/phone0|3") in new stack
-- Called phone0
-- Nobody picked up in 3000 ms
-- Hungup 'Phone/phone0'
-- Executing Playback("Phone/phone1", "vm/1234/unavail") in new stack
-- Created translator Phone/phone1/Translate:2
-- Destroyed translator Phone/phone1/Translate:2
-- Timeout on Phone/phone1
-- Executing Playback("Phone/phone1", "Goto") in new stack
*CLI> WARNING: File file.c, Line 362 (ast_streamfile): File Goto does not exist in any format
== Spawn extension (default, t, 1) exited non-zero on 'Phone/phone1'
-- Hungup 'Phone/phone1'
Thanks a lot.
Quote:
> *CLI> WARNING: File translate.c, Line 383 (translator_thread):
Blocking > > 'OSS/dsp', already blocked by thread 9226 in procedure
ast_write >
Quote:
Are you sure this is a full duplex sound card?
I'm thinking that it could be that it's doing a blocking write, and that
when the translator goes to read, the socket is already being written
to...
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