similar to: Asking

All the aforementioned techniques need change everytime on the dialplan. 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:

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,

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

> 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] > ... > allowguest=no >

Please can i ask you i want to know which code can help me to provide the taxation of voip/toip services in asterisk Le mar. 18 sept. 2018 à 01:36, Patrick Wakano <pwakano at gmail.com> a écrit : > Thanks everyone for the answers! > I did explored some options at the PHP level and probably will do > something in this direction, but in fact what I was really looking was >

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

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 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 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. 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

Hi guys. I am facing this problem here, using a2billing. error: 'Access denied for user 'a2billinguser'@'localhost' (using password: YES)' I am following this step by step http://www.asterisk2billing.org/cgi-bin/trac.cgi/wiki/Installation%20Guideand wend i get into the point that i have to Create a2billing database i am getting this message above. I even try to remove the

Hello I configured Asterisk to put CDRs in the database like it was explained in: www.voip-info.org/wiki/view/Asterisk+cdr+pgsql What I want to know is how do the billing solutions (like Asterisk2Billing) work with Asterisk. The billing system just use the information that Asterisk puts in the CDR table? Or they connect directly to Asterisk? Or is Asterisk that has, before the Dial command,

Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider) Now, with Asterisk2Billing would be something like this? exten => _2XXXXXXXX,1,Answer exten => _2XXXXXXXX,2,Wait,2 exten => _2XXXXXXXX,3,DeadAGI,a2billing.php exten => _2XXXXXXXX,4,Wait,2 exten =>

Hi list, I'm using Asterisk2Billing (v2.0.16) and it appears to have an annoying bug. When there are rates for e.g. 44 (UK landline) and 44870 (UK premium) and a fraudster manages to somehow dial 44-870 instead of 44870 the rate for 44 will match, not the one for 44870. So, I would like to block all calls on a dialplan level that contain a dash. -44, 4-4, 44-, 44---, -, ---, just

I couldn't run PostScriptTrace() from the package "grImport" without an error. At first the postscript program couldn't be found. however the problem persisted after the full path the postscript program was indicated. I read earlier post on the subject in vain. See the codes and output below. The file "Senegal_location_map.ps" was originally a "svg" file from:

I tried this [dongle0] ;audio=/dev/ttyUSB1 ; tty port for audio connection; no default value ;data=/dev/ttyUSB2 ; tty port for AT commands; no default value ; or you can omit both audio and data together and use imei=123456789012345 and/or imsi=123456789012345 ; imei and imsi must contain exactly 15 digits ! ; imei/imsi discovery is available on Linux only

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.

I'm sure this has been asked a million times. Therefore, I must ask again. Generally speaking, what do you guys think of it. It looks pretty good, but for my uses, I'm not sure that a calling card method is the *best* way to go. But, either way, what is the general concensus? -------------- next part -------------- An HTML attachment was scrubbed... URL: