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ok, that's really all I need to know. Of course, if anyone else wants to throw in their two cents, don't let me stop you :) -Thufir On Wed, Jul 6, 2016 at 1:36 AM, Frank Vanoni <mailinglist at linuxista.com> wrote: > I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with > Ubuntu Server 14.04. > > Works fine! :-) > > Frank > > On Wed, 2016-07-06 at 01:10 -0700, Thufir wrote: > > I'm debating between a cloud PBX or, perhaps, rasberry pi. For...

...I need the office secretary to edit a file (call file) and place it in a particular folder in their windows PCs. this folder is the outgoing folder of LINUX shared through samba in LAN. i need to make it as easy as possible, please. On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist at linuxista.com> wrote: > Here I'm using the "Page" application to make a conference call "on the > fly". > > > > [office] > > exten => 2222,1,Dial(SIP/desk2,150) > same => n,Hangup() > > exten => 3333,1,Dial(SIP/desk3,150) > same =...

...use imei=123456789012345 and/or imsi=123456789012345 ; imei and imsi must contain exactly 15 digits ! ; imei/imsi discovery is available on Linux only imei=352098043831724 ;imsi=123456789012345 My imei is 352098043831724 But nothing change. 2016-02-12 15:12 GMT-02:00, Frank <mailinglist at linuxista.com>: > On Fri, 2016-02-12 at 14:33 -0200, Vitor Mazuco wrote: >> Yes I used. >> >> The problem can be the version of Asterisk? >> >> I use Asterisk 13 instead of 11. > > Try > > [dongle0] > imei=347654458453667 > imsi=976895757545778 > >...

On Tue, 2016-02-23 at 17:06 +0000, Steve Howes wrote: > Google?... Yeah... searched "google voice recognition api asterisk", browsed though various results. Nothing helpful for a beginner, very confusing bla bla... Thanks anyway for your help. F.

> On 11/02/2017, at 3:40 am, Frank Vanoni <mailinglist at linuxista.com> wrote: > > On Thu, 2017-02-09 at 14:58 +0200, ????? ?????? wrote: > > >> so the main question is -- how to Disallow CALLS without registering >> on PBX > > sip.conf configuration > In the [general] section, define: > > > [general] > ... &gt...

Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA&SIP/deviceB&SIP/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension to dial in order to modify the dialplan. Here is what I

HI ALL got small question i use call-limit=1 on peers but call limit is not working if user is not registered on PBX and making calls so the main question is -- how to Disallow CALLS without registering on PBX -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy

Hello, I am looking for a replacement for my Thomson ST2030SIP. My specifications are as follows : - 2 lines. - 6 BLF keys. - PoE. Can you give me a return on the models you use ? Thanks. Sil

Hi Daniel On Mon, 2016-02-22 at 19:40 +0100, Daniel Heckl wrote: > try this http://zaf.github.io/asterisk-speech-recog/. > I have tested it myself, it works very well. I wanted to try it, but I obtain the following error message: "speech-recog.agi,en-US: API key is missing. Aborting. " :-( What am I missing? Any hint? Frank

On Tue, 2016-02-23 at 00:43 +0100, Laszlo wrote: > ------------ > Requirements > ------------ > ....... > Speech API key from Google Yes... OK... but... where and how can I obtain this API Key? Where and how do I install it into my Asterisk box?

Hi to everybody, My IAX is not working, When I type reload IAX it returns me: AsteriskSlave*CLI> iax2 reload == Parsing '/etc/asterisk/iax.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13546 set_config: Ignoring bindport on reload [Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13610 set_config: Ignoring bindaddr on

Hello, please can i have a code which help me to tax user every voip services in asterisk means when user starts to call someone -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180918/b2a02a8b/attachment.html>

VoipRaider the site, says calls to landlines in Brazil is FREE within the freedays period. Log in to the website and hire the service, it says that I have 90 days of freedays paying for cheaper service is $ 10.. That is from what I understand, I will pay 10 dolares for unlimited call in landlines for a period of 90 days? Is that it? Has anyone tested it there? How many simultaneously calls can

Hi, I have already tried to change for bindaddr=0.0.0.0 but it didn't worked. 2016-08-26 11:44 GMT-03:00, Frank Vanoni <mailinglist at linuxista.com>: > On Fri, 2016-08-26 at 10:12 -0300, Vitor Mazuco wrote: > >> bindaddr = all > > Try: > > bindaddr=0.0.0.0 > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www....

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

Hi everyone! I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but my Huawei E153 is not working in my Asterisk. I fallow this rules http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14 But not successes. Thanks in advanced,

I'm debating between a cloud PBX or, perhaps, rasberry pi. For a SOHO, maybe three hardphones, rasberry pi would suffice? I would be amazed, but, if so, great. thanks, Thufir -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160706/84a0ac7c/attachment.html>

Hi. in my system i have a conference room where someone can call it eg 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in through a different number and PIN. I would like to have a call file and call all participants eg 610-619 at certain time of the day and give them access to the conference. During my try i managed to create a call file where it calls the a SIP phone and

Yes I used. The problem can be the version of Asterisk? I use Asterisk 13 instead of 11. 2016-02-12 14:31 GMT-02:00, Shabbir abbasi <shabbirabbasi92 at gmail.com>: > have changed this > [dongle0] > audio=/dev/ttyUSB1 > data=/dev/ttyUSB2 > > To > > [dongle0] > imei=123456789012345 > > and imei exact same as on your device ? > > On Fri, Feb 12, 2016

On Sat, 2020-10-03 at 22:25 +0200, Sebastian Nielsen wrote: > many providers in sweden have started disabling SIP account details > and now require usage of their own ”router’s”. That's very irritating and make me angry. Few of my client had the same problem. The solution: write a letter asking the SIP credentials explaining you want configure your own equipment and tell them you