asterisk users - Nov 2004 - Grandstream BT100 - No Sound with Playback()

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 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
;mailbox=1234@default  ; 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

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
;mailbox=1234@default  ; 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