It works fine for me, I created a 2nd "music on hold", tossed a bunch of mp3 files into a directory and I can listen to music on the speakerphone:
;radio @ 8888 exten => 8888,1,Answer exten => 8888,2,MusicOnHold(default) > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Steve Haehnichen > Sent: Sunday, September 14, 2003 1:14 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] MusicOnHold (MOH) silent on BudgeTone-100 only. > > pamAssassin 2.55 (1.174.2.19-2003-05-19-exp) > > I have the MusicOnHold feature working great when called from ATA-186 > extensions. It's pretty cool. > > However, when I call from a BudgeTone-100 phone, no music is heard -- > instead it continues the ringing feedback and acts like the call is > unanswered. At the same time, I can call from (multiple) ATA-186 > extensions and hear music as long as I like. How can I debug this? > As far as I can tell, Asterisk thinks it's connecting the BudgeTone > channel just fine. Only it's silent. > > Normal voice calls between BudgeTone and ATA-186 extensions work fine, > of course. They use ulaw encoding. It's just the MOH that fails to > answer. There is no NAT or firewall involved -- all hosts are local. > > I'm testing with dedicated extensions like this: > > exten => 302,1,WaitMusicOnHold(5) > exten => 303,1,MusicOnHold(loud) > exten => 304,1,MusicOnHold(default) > > Here's the asterisk output from a working ATA-186 call: > *CLI> > -- Executing MusicOnHold("SIP/200-990d", "loud") in new stack > -- Started music on hold, class 'loud', on SIP/200-990d > > [ beautiful music emanates from earpiece ] > > -- Stopped music on hold on SIP/200-990d > == Spawn extension (dialout, 303, 1) exited non-zero on 'SIP/200-990d' > > > And from the silent BudgeTone-100: > -- Executing MusicOnHold("SIP/202-351f", "loud") in new stack > -- Started music on hold, class 'loud', on SIP/202-351f > > [ Budgetone still provides ringing indication until I end.] > > -- Stopped music on hold on SIP/202-351f > == Spawn extension (dialout, 303, 1) exited non-zero on 'SIP/202-351f' > > > While the BudgeTone is 'ringing', Asterisk appears to think it has a > live connection: > > *CLI> sip show channels > Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter > Format > 192.168.42.56 202 26e8973e-e3 00101/06079 00000ms 0000ms > ULAW > 1 active SIP channel(s) > > Has anyone else tried MOH from a Grandstream BudgeTone extension? It > would mean a lot just to hear that it's my local config and not some > weird BT problem. If someone could post a working BudgeTone SIP > definition, that might help too. > > Thanks! > -Steve > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users