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I've been noticing that my group of Polycom IP 501 phones seems to randomly reset themselves nearly every night (I guess it usually happens at night, since I've never seen it happen while I've been at work during the day).. When I say "reset", I mean, the hands free volume and ring volume are set to the default and the call logs (received calls, missed calls, placed calls)

Can anyone tell me how me to get asterisk to dial out a phone number when a bluetooth device is not detected? I've tried putting the following under the clients section in /etc/asterisk/btp.conf: client =>user,00:12:34:56:78:90,Zap/4/1234567890 and in extensions.conf: exten => 222,1,Playback(pls-hold-while-try) exten => 222,2,Dial(BTP/user,60,m) exten => 222,3,Hangup but

I sent the following message a few days ago, but never received a reply, so I thought I'd ask again.. Can anyone tell me how me to get asterisk to dial out a phone number using BTP when a bluetooth device is not detected? I can get BTP to dial to a SIP phone, but I can't get it to dial through a POTS phone line using the Zap interface.. I've tried putting the following under the

I've just set up my dial plan to use hints for each of my extensions, and I've set up a buddy watch list on my Polycom IP-501's, however, I keep getting the following messages every 5 or so minutes: -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.1.106 -- Incoming call: Got SIP response 500 "Internal Server Error" back from

I've tried posting a simple message twice, regarding using chan_btp to dial a phone number through a Zap interface, but I've received no answers, and I can't seem to figure out how to do what I want (which seems to be a pretty typical use of chan_btp - I mean, isn't it used to dial peoples' phone numbers when their bluetooth device is not present?). I was wondering if anyone

I've been using Asterisk 1.2.6 with a 4 port FXO Sangoma A200 card for the past month without many problems (other than the fact that the Sangoma card doesn't disconnect hung up calls immediately, which I posted about in another thread, and has still not been fixed), however, I had a call from one of our clients today and they complained that our phone system kept ringing through when they

When someone calls into our asterisk server over a PSTN line, dials an extension and then hangs up, the SIP phone related to the given extension will ring about 4 or 5 times before asterisk shows that the channel has been hung up in the console. This isn't such a big deal on its own, but what's happening now is that if a user calls in from a PSTN line, gets voicemail on the extension, and

As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing