similar to: Bluetooth (chan_btp): dialing external phone number through BTP/Zap when bluetooth device not present?

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'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

hi, we have successfully implemented at tang/clevis environment for automatically entering luks keys and booting hosts without operator intervention. Now we would like to use this as well on ipv6 networks, but I do not seem to get it to work. I have already posted this issue to the dracut devs github issue tracker ( https://github.com/dracutdevs/dracut/issues/554) but no response so far. Maybe

Hi, I am new to Asterisk and I am having a setup problem that I am trying to resolved for the last couple days without any success. I am pretty much desperated on this issue and I don't know why. Can someone please kindly help me to troubleshoot this? I can't hear any audio from Asterisk when running Playback or VoiceMail tests. I have my Asterisk server ( running on Debian,

Dear R-friends, the dataset I am using (data.it) is organized as follows partner stp btp reg hk 0.64 1 s ger 0.27 1 d tur 0.27 1 s rom 0.24 1 s-f por 0.24 1 s spa 0.23 1 s gre 0.22 1 d-f aus 0.17 1 d uk 0.16 1 s be 0.16 1 d arg 0.15 1 s usa 0.13 1 d-f fra 0.13 1 s neth

Just to make clear how ridiculous this has gotten [Not RP's position, but the "opposition"] let me put the demands in context. I'll just restate them so they are clearly as entitled and crazy as they really are. >> Ubuntu can recommend what they like, but Samba only recommends and >> supports ntpd. RHvs> no, you think so, you even maybe right but all you do in

Hi, I am using cwrsync ver 2.1.5 over SSH to connect to myproj@sourceforge.net and facing some problems and not sure how to go about it. Below is the full text of the result when I run the batch file. Was wondering if anyone has seen any similar issue. Any thoughts/help is much appreciated. ------------------------------------------------ Tunnel: ssh Command to run: "C:\Program

Hi All Does anyone know if one could use bluetooth on a cell phone with *? Would be nice to have your cell as an office phone combo. I heard that there is a bluetooth module for *? If so this should be possible? Thanks Doug --- NOTICE - This message contains privileged and confidential information intended only for the use of the addressee named above. If you are not the intended recipient

Hi, I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the box (in a hope to sort out the uname -r issue mentioned below). I'm using the Asterisk Doc Proj vol 1 to guide me through the initial setup. I have no special HW and intend to use asterisk on an internal network just to get some experience. I have downloaded what I think I need and placed it in /usr/src (see

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

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)

Hi, I have a problem with my IAX softphones. After a call, when the softphone hangup, it remains unavailable for the other softphones. It can call anybody, but can not be reached... For example, if A call B, B answer, then A or B hangup, and C won't be able to call A or B after that (but A or B would be able to call C). The Dial function returns that the chan is unavailable. That is very

Hi Stephen, it's me again with a problem report about quotas and ext3... I'm not quite sure whether you tried to resolve the quota problems yet, but since I read in the changelog of 0.0.5e that you included some quota specific fixes I decided to give a try (again). And once again: Bad news. Still lockups, reliable reproducible. But this time I have a little backtrace for you of two

Hi folks, Anyone want to check out the subject packages will find them here: ftp://www.linuxsys.com/pub/testing/asterisk The asterisk RPMS now include builds of asterisk, addons, astcc, btp, gastman, iaxproxy, libpri and zaptel from 10.04.04 CVS. zaptel is compiled and working with kernel-2.4.22-0.FC1.LSE.p3.i686.rpm available with ALSA modules and supporting packages here:

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

hi , all what's wrong with this command? exten => 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1) as i got the error: -- Executing [222 at 95040:1] GotoIfTime("SIP/1001-00000099", "11:00-14:00|mon|wed|*|*?1:3|1") in new stack [Jan 20 11:21:11] WARNING[16804]: pbx.c:4118 get_range: Invalid day 'wed', assuming none but what should i do. if i want to set

Good day, I've got around to installing an X100P card in my computer to try out asterisk. I noticed (and people who were testing with me also noticed) that when dialing from my SIP soft phone to the PSTN, the ringer tone changes after 2-3 seconds, precisely when the Zap channel takes over the call. Is it possible to eliminate the first ringing? Is there a reason to this

I'm using a T100p to interface to a channel bank and from there to analog PSTN lines. Because of my particular setup I have to do post-connect inband DTMF dialing - which takes up to 5 seconds for a 10 digit number, assuming 0.5/sec per digit (ie. using "zap/g1/31|5|D(6045551212)". Even with an 'outside transfer' voice prompt before commencing dialing my users are getting

> -----Original Message----- > From: Adam Goryachev [mailto:mailinglists@websitemanagers.com.au] > Sent: September 7, 2004 8:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Monitored outbound dialing via Zap > interface? {clip} > Have you considered adding the r option to the Dial command, so they > might hear ringing

Hi, After using Asterisk with a SIP hardphone for a couple of weeks I've just installed a TDM400P card. My hardphone - a 7940 - allows me to use a dialplan to decide when a particular extension is complete and automatically trigger dialing. This works well with my internal extensions, which are all of the form "Z00". When trying to dial these extensions from a handset