Hi Everyone,
I'm having a problem with a Grandstream Budge Tone 100 phone. When
Asterisk send sound to the extension using Playback I'm getting the
following message:
-- Executing Playback("SIP/2002-01fe", "tt-monkeysintro") in new stack
-- Playing 'tt-monkeysintro' (language 'en')
Nov 11 11:54:02 WARNING[278540]: file.c:550 ast_readaudio_callback: Failed
to write frame
== Spawn extension (from-sip, 5555, 1) exited non-zero on 'SIP/2002-01fe'
I tried every codec on the phone with no succes...
Here is the entry for the Grandstream phone in my sip.conf
[2002]
type=friend ; either "friend" (peer+user), "peer" or
context=from-sip
username=2002 ; usually matches the section title
fromuser=2002 ; overrides the callerid, e.g. required by FWD
secret=123456
callerid=John Doe <2002>
host=dynamic ; we have a static but private IP address
nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
dtmfmode=rfc2833 ; either RFC2833 or INFO for the BudgeTone
;[EMAIL PROTECTED] ; mailbox 1234 in voicemail context "default"
;allow=all ; need to disallow=all before we can use allow
disallow=all
allow=ulaw
The BT100 and Asterisk are on the same lan... It's look like every time
the playback function is executed the BT100 just hangup.
Thanks for your help,
Dave
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