I receive this message when I pick up my phone plugged to PhoneJack card
and I receive a busy tone.
== Accepting call on 'Phone/phone0'
-- Executing Playback("Phone/phone0", "demo-abouttotry") in new
stack
*CLI> WARNING: File file.c, Line 405 (ast_streamfile): Unable to find a
translator method
== Spawn extension (default, 500, 1) exited non-zero on 'Phone/phone0'
-- Hungup 'Phone/phone0'
Any idea???
--
==============================================================================
Ip6 Seguridad S.L jddr@ip6seguridad.com (Jorge de Diego Rodriguez)
Departamento VozIP
<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
I receive this message when I pick up my phone plugged to PhoneJack card
and I receive a busy tone.
<br>
<br>
<p> == Accepting call on 'Phone/phone0'
<br> -- Executing Playback("Phone/phone0", "demo-abouttotry")
in new stack
<br>*CLI> WARNING: File file.c, Line 405 (ast_streamfile): Unable to find
a translator method
<br> == Spawn extension (default, 500, 1) exited non-zero on 'Phone/phone0'
<br> -- Hungup 'Phone/phone0'
<br>
<pre></pre>
<pre>Any idea???</pre>
<pre></pre>
<pre>--
==============================================================================
Ip6 Seguridad S.L jddr@ip6seguridad.com (Jorge de Diego Rodriguez)
Departamento VozIP
Posted: Thu Mar 30, 2000 5:45 pm Post subject: [Asterisk] Asterisk+PhoneJack
On Thu, 30 Mar 2000, Jorge de Diego Rodriguez wrote:
Quote:
I receive this message when I pick up my phone plugged to PhoneJack card
and I receive a busy tone.
== Accepting call on 'Phone/phone0'
-- Executing Playback("Phone/phone0", "demo-abouttotry") in new
stack
*CLI> WARNING: File file.c, Line 405 (ast_streamfile): Unable to find a
translator method
== Spawn extension (default, 500, 1) exited non-zero on 'Phone/phone0'
-- Hungup 'Phone/phone0'
Any idea???
You're probably communicating with Asterisk using g723.1. (And you don't
have a codec for it.) ommunicate using slinear mode.
Posted: Fri Mar 31, 2000 2:39 pm Post subject: [Asterisk] Asterisk+PhoneJack
Quote:
== Accepting call on 'Phone/phone0'
-- Executing Playback("Phone/phone0", "demo-abouttotry") in new
stack
*CLI> WARNING: File file.c, Line 405 (ast_streamfile): Unable to find a
translator method
== Spawn extension (default, 500, 1) exited non-zero on 'Phone/phone0'
-- Hungup 'Phone/phone0'
The card is in G.723.1 format, and because of the proprietary nature of
G.723.1 I am unable to provide a translator into singed linear format,
thus allowing translation to occur from the format of all the demo files.
You have two choices:
1) In /etc/asterisk/phone.conf, tell the phonejack to use slinear format
instead of g.723.1 -- I've found this to be soemwhat unstable, but I have
an older driver as well. Of course, this defeats the compression that
you've paid $$ for in purchasing the phonejack, but oh well.
2) Record your own messages in G.723.1 format, and copy them into the
/var/lib/asterisk/sounds with the same names (except the extension
".g723") and Asterisk will know to use them.
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