davis@kangunet.net
2004-Nov-11 10:56 UTC
[Asterisk-Users] 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
davis@kangunet.net
2004-Nov-11 11:00 UTC
[Asterisk-Users] 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 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