Re: [asterisk-users] Mailing List Future

On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: > To that end, we’ve decided to discontinue the mailing lists effective > February 1st, 2024. That's a very sad news! :-( -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] audio from soft phone actual phone from cloud

On Wed, 2023-07-19 at 12:42 -0400, Jerry Geis wrote: > Why might I not be getting audio ? Make sure the RTP port range is correctly configured and open on your server's firewall. The port range is defined in /etc/asterisk/rtp.conf The same range of UDP ports must be correctly forwarded on your

Re: [asterisk-users] Intro and question

Even I was confused, and the directions in that book seem like a complication of a simple affair, at least for my modest needs. Finally, I installed Asterisk with apt and created extensions.conf and pjsip.conf files. -- _ --

Re: [asterisk-users] Dial() after the h extension has been invoked?

On Fri, 2021-11-12 at 16:56 +, Antony Stone wrote: > I use Dial() commands with custom SIP headers to pass information > (eg: about > the current state of a call) between the front-end and back-end > machines, and > this works very well. > > I need to perform a Dial() > command after an

Re: [asterisk-users] Notifying missed calls

On Sat, 2021-11-06 at 14:46 +0100, Luca Bertoncello wrote: > Really, I can't understand what you mean... I'm feeling really > dumb... No need to feel dumb. I'm not an expert and when I look to my extensions.conf... well... countless pulling my hairs out, head banging on the keyboard,,, :-) The

Re: [asterisk-users] Notifying missed calls

On Fri, 2021-11-05 at 10:50 +0100, Luca Bertoncello wrote: > 1) The E-Mails will be sent "double" It sends the first mail by executing "noanswer,2" and a second mail because because of "main-incoming,h,2" > 2) The E-Mails will be sent for outgoing unanswered calls, too. Use the "h" extension

Re: [asterisk-users] Notifying missed calls

Here my configuration: [incoming] ; Incoming from Swisscom exten => +4191xxx,1,NoOp(Call from ${CALLERID(num)}) same => n,Dial(SIP/deskphone,120) same => n,Hangup() exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done) exten => h,n,System(echo "Missed Call from ${CALLERID(num)}" |

Re: [asterisk-users] Detect if people is talking

On Wed, 2020-12-30 at 12:09 -0300, Valter Nogueira wrote: > Is there any way to detect if an agent is speaking? https://wiki.asterisk.org/wiki/display/AST/Application_WaitForSilence https://wiki.asterisk.org/wiki/display/AST/Application_WaitForNoise --

Re: [asterisk-users] which linux for asterisk?

On Wed, 2020-12-09 at 11:03 +0400, Dmitry Melekhov wrote: > what is best choice ? Oracle? Ubuntu? I'm running Asterisk since several years on Ubuntu without any issues. Debian should be fine too. -- _ -- Bandwidth and

Re: [asterisk-users] Anyone that know of DECT "client" for asterisk?

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

Re: [asterisk-users] Mail2Fax

On Wed, 2020-06-17 at 18:10 +0200, basti wrote: > txfax seem to be a port of spandsp. it is also old. > Is there a newer way to send fax via asterisk. I don't know if it's newer, but I use "sendfax" -- _ -- Bandwidth and

Re: [asterisk-users] email notification on missed call

On Sat, 2019-11-02 at 11:42 +0100, Antony Stone wrote: > Doesn't that send an email for every call once it ends, not just > unanswered ones? Whoops! You are right! :-) exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done) exten => h,n,System(echo "Missed Call Open on Asterisk from

Re: [asterisk-users] email notification on missed call

On Wed, 2019-10-30 at 05:10 +0100, Fourhundred Thecat wrote: > what is the best way to implement email notification on missed call ? > Is there perhaps a better way to this than described above ? This is my way: exten => h,1,System(echo "Missed Call Open on Asterisk from ${CALLERID(num)}" |

Re: [asterisk-users] Asterisk 13.28.0 Now Available

Thank you, dear Asterisk Development Team, for this great software! > The Asterisk Development Team would like to announce the release of > Asterisk 13.28.0. -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Sending SMS and SIM card

On Fri, 2019-04-26 at 14:39 +, bilal ghayyad wrote: > Any small example how to send gsm calls through chan_dognle and how > to send sms through chan_dongle? To send SMS, there is a CLI command. You can use the commands in your extensions.conf accordingly your needs.

Re: [asterisk-users] Sending SMS and SIM card

On Fri, 2019-04-26 at 14:39 +, bilal ghayyad wrote: > Any small example how to send gsm calls through chan_dognle and how > to send sms through chan_dongle? In dongle.conf: [gsmgateway] context=gsm imei=123456789012345 imsi=098765432112345 In extensions.conf: [gsm] ; Incoming calls from

Re: [asterisk-users] Sending SMS and SIM card

You can use a cheap 3G-USB-dongle and chan_dongle. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk?

Re: [asterisk-users] Asking

On Tue, 2018-09-18 at 20:28 +0200, modou lo wrote: > 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 Check Asterisk2billing  http://www.asterisk2billing.org/ --

Re: [asterisk-users] Decoding SIP register hack

On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote: > 3. How do I set up the server to block these ? > > 4. Can I stop the retransmitting of the 401 Unauthorized packets ? I'm happy with Fail2Ban protecting my Asterisk 13. Here is my configuration: in /etc/asterisk/logger.conf: messages =>

Re: [asterisk-users] invite to conference by a call file

Maybe something like a local web page where your secretary can enter the list of phone numbers to call and a script that generates a call file and moves it to the Asterisk spool folder. But that's not an Asterisk issue. It's more a programmer's issue. :-)  On Thu, 2018-03-22 at 16:06 +0200,

Re: [asterisk-users] invite to conference by a call file

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@linuxista. > com> wrote: > > Here I'm using the "Page" application to make a conference call &quo

Re: [asterisk-users] invite to conference by a call file

Here I'm using the "Page" application to make a conference call "on the fly". [office] exten => ,1,Dial(SIP/desk2,150)    same => n,Hangup() exten => ,1,Dial(SIP/desk3,150)    same => n,Hangup() exten => ,1,Dial(SIP/desk4,150)    same => n,Hangup() exten =>

Re: [asterisk-users] Blacklist failed attempts

On Thu, 2018-03-01 at 15:02 +0200, Atux Atux wrote: > I have tried to implement it through fail2ban, but it doe snot seem > to work for my asterisk implementation. I'm happy with Fail2Ban protecting my Asterisk 13. Here is my configuration: in /etc/asterisk/logger.conf: messages =>

Re: [asterisk-users] email when certain numbers are called

On Mon, 2018-01-15 at 14:26 +0200, Atux Atux wrote: > [DefaultPlan] exten => _XX,1,System(echo "Dialed number ${EXTEN} on Asterisk from ${CALLERID(num)}" | mail -s "Dialed number ${EXTEN} on Asterisk from ${CALLERID(num)}" -a "From: Asterisk PBX " yo

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

> fail2ban is most useful for blocking registration attempts.    I > handle  > non-registration call attempts by allowing guests, point them to a > jail  > context, which runs Log(WARNING,fail2ban='${CHANNEL(peerip)}')   I > set a  > fail2ban rule to match that line logged from Asterisk. Thanks

Re: [asterisk-users] SIP trunks going to the wrong context

I don't know if it applies to your problem, but I also had some troubles with multiple account on same SIP provider.  Here what works for me: In sip.conf: register => 11:qwe...@sip.provider.zz/11 ; Trunk1 register => 22:asd...@sip.provider.zz/22 ; Trunk2 register =>

Re: [asterisk-users] Looking for the carrier that owns a particular DID

On Thu, 2017-11-02 at 11:33 -0400, Tech Support wrote: > How do I find out which carrier owns the DID in question? Try here: https://www.twilio.com/lookup -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] PJSIP, NAT and STUN/ICE

On Tue, 2017-10-10 at 11:32 +0200, Frank Vanoni wrote: > On Mon, 2017-10-09 at 23:56 +0200, O. Hartmann wrote: > > > > > local_net=  192.168.254.1/24 > > It should be: > > localnet = 192.168.254.0/255.255.255.0 Whoops.

Re: [asterisk-users] PJSIP, NAT and STUN/ICE

On Mon, 2017-10-09 at 23:56 +0200, O. Hartmann wrote: > local_net=  192.168.254.1/24 It should be: localnet = 192.168.254.0/255.255.255.0 -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Dial an extension to modify dialplan

On Wed, 2017-05-10 at 12:56 +0200, Frank Vanoni wrote: > exten => 2001,1,Dial(SIP/Dial(SIP/deviceA/deviceB/deviceC) > > exten => 2002,1,Dial(SIP/Dial(SIP/deviceA/deviceB) Whoops... sorry for the typo (in the hurry of copy & paste)! exten => 2001,1,Dial(SIP/deviceA/d

Re: [asterisk-users] Dial an extension to modify dialplan

Dear Digium List First of all, I thank all of you for all the replies and the interesting suggestions. I thank you very much. I can only learn from people like you. :-) I will remember all the different solutions for a future use in other scenarios. On Mon, 2017-05-08 at 16:35 +0200, Frank

[asterisk-users] Dial an extension to modify dialplan

Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA/deviceB/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

Re: [asterisk-users] log incoming calls without answering

On Thu, 2017-04-20 at 17:26 -0300, Fabio Moretti wrote: > Any idea? I used to play with an analog telephone line and Asterisk by using a Linksys SPA-3102 Voice Gateway. I think it is no longer manufactured, but maybe you con buy a used one on eBay or you can find an equivalent device from

Re: [asterisk-users] Disallow CALLS without registry

On Sat, 2017-02-11 at 12:25 +1300, Pete Mundy wrote: > > sip.conf configuration > > In the [general] section, define: > > [general] > > ... > > allowguest=no > > alwaysauthreject=yes > > ... > With the above configuration on my Asterisk, I obtain the following result: - if the phone is

Re: [asterisk-users] Disallow CALLS without registry

On Thu, 2017-02-09 at 14:58 +0200, Антон Сацкий 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 alwaysauthreject=yes ... The "allowguest" line disables anonymous SIP

Re: [asterisk-users] MOBILE SIMCARD ON ASTERISK

Hi Chris On Tue, 2016-12-06 at 04:36 +0200, christopher kamutumwa wrote: > Is it possible to have a simcard configured and become incoming line > and outgoing on asterisk and also have the IVR function? Yes, it is possible! :-) A cheap solution is using a 3G-UBS-dongle. I have two SIM cards

Re: [asterisk-users] missed call notification

On Mon, 2016-11-28 at 14:31 +0100, tux john wrote: > Hi. i am running asterisk 11 in debian and i would like have a missed > call notification down to extension level. > so if i get a missed call to extension 6589 then send an email to the > user's email address with a subject and a text message.

Re: [asterisk-users] Blacklist callers from file

On Sat, 2016-08-27 at 17:59 +0200, tux john wrote: > Hi. I would like to blacklist a few callers Example: callers with CallerID 0123456789, 9876543210 and 7410258963 are sent to tt-monkeys. Callers from area code 555 are also blocked. In "extensions.conf" file add #include "blacklist.conf"

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

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.api-digital.com -- Join the Asterisk Community at the 13th AstriCon,

Re: [asterisk-users] rasberry pi

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 a > SOHO, maybe three hardphones, rasberry pi would suffice? I

Re: [asterisk-users] Including doesn't have any effect

On Mon, 2016-06-06 at 08:08 -0700, Steve Edwards wrote: > The purpose of a subroutine (code that is entered by a gosub and exited by > a return) is to allow the creation of easily reusable code. [snip] Steve Thank you very, very much for your answer. I really appreciated your interesting

Re: [asterisk-users] Including doesn't have any effect

On Mon, 2016-06-06 at 17:47 +0100, Julian Beach wrote: > exten => s,n,GotoIf(${DB_EXISTS(blacklist/${CDR(src)})}?block) ; Check > whether caller blacklisted As far as I know, Asterisk's database/blacklist function only supports exact match of caller ID. If you want to block a specific area code

Re: [asterisk-users] Including doesn't have any effect

On Sat, 2016-06-04 at 15:19 -0700, Steve Edwards wrote: > Using a 'goto' to exit from a gosub is a bad idea. Why? > A better idea would be > to set a channel variable and check it's value after the return, in the > calling context. The idea is to update the blacklist.conf whenever I want to

Re: [asterisk-users] Including doesn't have any effect

Another possible approach to blacklist two (or more) specific callers (098765432 and 012345678 as example) In extension.conf #include "blaklist.conf" exten => _+x.,1,Gosub(blacklist,s,1) exten => _+x.,n, exten => black,1,playback(tt-monkeys) In blacklist.conf exten =>

Re: [asterisk-users] How to set outgoing sip callid ?

In sip.conf [devicename] callerid="Jon Doe" <+123456789> or in extensions.conf exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) exten => 1234,n,Dial(SIP/ -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] VoipRaider is true for FREE calls?

On Mon, 2016-05-09 at 19:43 -0300, Vitor Mazuco wrote: > VoipRaider the site, says calls to landlines in Brazil... I hope I'm not infringing any mailing list rule by recommending you to take a look to the following providers. I use them with my Asterisk, the rates are good and they allow

Re: [asterisk-users] VoipRaider is true for FREE calls?

VoipRaider is a service from DELLMONT SARL. This company offers voip services under dozens of different domains (voipcheap, voipdiscount, onevoip,...) Search "Dellmont Sarl" in Google and read the user's reviews. Personally, I would never send a penny to them... Franky --

Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

Just a few ideas... 1. Disable all mobile carrier's voicemail and configure a voicemail on your Asterisk. Let Asterisk handle the unanswered calls. 2. If your SIP provider allows multiple calls at the same time, configure Asterisk to call all your SIMs at once (instead of calling the first,

Re: [asterisk-users] How to install Huawei E153 in a Asterisk 11 or 13?

On Wed, 2016-03-02 at 19:12 -0300, Vitor Mazuco wrote: > I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but > my Huawei E153 is not working in my Asterisk. > But not successes. A little more information from you would be helpful to identify the problem. I have a Huawei USB

Re: [asterisk-users] Handle a call if one phone of a ring group is busy

On Sun, 2016-02-28 at 01:43 +0100, Frank wrote: > Question: How to give a "busy signal" back to the caller if one > extension of a ring group is in use? Or redirect the call to voice mail? Found a solution! :-) exten => 7654321,1,GotoIf($["${DEVICE_STATE(SIP/111)}"="INUSE"]?Busy,1) exten =>