search for: db_exist

...g variable value won't be the expected one (a string) but some kind of number However this work: Set(queueMemberList=${QUEUE_MEMBER_LIST(${queue})}); Is there a reason ? i have the same comportement with CUT() Also, there is this construct which isn't working: if( ${DB_EXISTS(family/key)} ) { } else { } However i can't find a workaround in this case ... any idea ?

...y NAT translation alive. Anyway, here's the dialplan for the IVR (only extensions 302 and 303 are included for brevity): [ivr-XXXXXX] exten => s,1,Answer exten => s,n,Playback(silence/1) exten => s,n,Background(XXXXXX/greeting) exten => s,n,WaitExten(4) exten => 302,1,GotoIf(${DB_EXISTS(CFIM/302)}?dialfw:dial) exten => 302,n(dialfw),Set(extension=${DB(CFIM/302)}) exten => 302,n,Set(wait=${MATH(${DB(NumRing/302)}*6,int)}) exten => 302,n,ExecIf($["${wait}" != "0"]|Dial|SIP/XXXXXX2|${wait}|g|) exten => 302,n,Dial(DAHDI/g1/${extension},90,g) exten =&gt...

...erID. I created an asterisk database and with entries in the database for all extensions in the second office and defined the following macro: globals { CONSOLE="Console/dsp"; TRUNK="Zap/r1"; TCTC_Operator=15; Law_Operator=12; }; macro outside-dial ( num ) { if (${DB_EXISTS(Office/${CALLERID(num)})}) { TRUNK="Zap/r2"; } else { TRUNK="Zap/r1"; } Dial(${TRUNK}/${num},,Ttok); } It's working and correctly routing outside calls, but I get the following messages when I reload the extensions.ael file: [Dec 23 12:16:22] WARNING[2994]:...

Hi, How can I strip + from the front of the caller ID? I have tried this: exten => s/_+X.,1,Set(CALLERID(name)=${CALLERID(name):1}) But it is not working. Szasz Szabolcs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100114/a3f18781/attachment.htm

Hi I have to setup call forwarding. How do we setup Call forwarding in asterisk?. Eg. user dials a number and insert some mobile number for forwarding and dial another number to cancel the forwarding. thanks a lot. Best Regards, Madushan? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

...39;re > allowing it for all that can reach the context) > > The divert{off,active,void} subroutines are where I handle the > announcements - but you could probably easily implement your own. > > At the top of my [voicemail] context, I do this: > > > exten=>s,1,GotoIf(${DB_EXISTS(divert/${ARG3})}?outbound-standard,${DB_RESULT},1) > > (ARG3 contains the last four digits of the number the call came to in > my case, and a success passes the call via the "outbound-standard" > context which is in my dialplan. Your exact requirements may vary but > this...

...x > samples that might be of use in implementing it. > > They are checking a DEVICE_STATE to see if a phone is BUSY, You could > change it to be a database call or implement custom device states and check > those. > > wrapping your test case in an ExecIF statement that uses the DB_EXISTS > function to see if the database field you are checking is valid so you > don't get errors about non existent entries. > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB_EXISTS > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB > &g...

Hello Doug, Friday, December 13, 2019, 11:03:37 AM, you wrote: >> This is exactly what I do - “press 1 for a human” >> Works great > I do this as well, but I also do a database lookup to see if the number > is on our speeddial list and if so, pass the call directly on without > the IVR prompts. I do something similar for calls without caller ID, but I was still getting

Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same. Any ideas on how to overcome this problem as we dial

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please

Hi I have setup forwarding - xfering - where you press # and then the extension. I add t to the dial cmd. My problem is that when you call something like internet banking they want #, but when # is pressed asterisk gets it instead. is there a way around this ? I haven't been able to get asterisk to listen to flash either Alex -- Why is it there are so many more horses' asses than

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan

In my blacklist I have: database show ... /blacklist/Manitoba : advertising ... [incoming] ; First, lets take care of telemarketers exten => 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1) exten => 4,n,Set(goaway=${CALLERID(number):0:2}) exten => 4,n,GotoIf($["${goaway}" = "V4" ]?blacklisted,s,1) exten => 4,n,GotoIf($["${goaway}" = "V3"

..., but not only does Confbridge not return, but i'd need something that erases the database entry after the conference is empty, not after 1 particular user leaves. [macro-confbridge-setup] exten => s,1,Set(NUM=$[0${NUM} + 1]); exten => s,n,Set(CONFNO=99${NUM}) exten => s,n,GotoIf(${DB_EXISTS(confbridge:${CONFNO})}?1) exten => s,n,Set(DB(confbridge/${CONFNO})=1) [foo] exten => s,1,Macro(confbridge-setup) exten => s,n,ConfBridge(${CONFNO}) exten => s,n,NoOp( ${DB_DELETE(confbridge/${CONFNO})} ) -- Jeremy Kister http://jeremy.kister.net./

Hi, I have been looking for an example of accomplishing this, but I've been unable to locate something similar to what I'm trying to do. Here's the scenario: Users caller ID is set to their internal extension (200-250). This is set in sip.conf for each user. Each user has a local DID as well (hosted through Vitelity, for example (555)111-2222). The problem is that this extension was

> 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 >

That is the general idea. But how do i make it work? is there somewhere ready? On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkiniston at gmail.com> wrote: > Define your *72 and *73 extensions in your internal context, Have them set > a value in the ASTDB that you then check when dialing your handsets. > > The same can be done for call forwarding, store a number in the

Hi AJ, Am 06.06.2016 um 10:14 schrieb A J Stiles: > But why not call an AGI script, have this check the caller ID against a MySQL > database and return a status -- blocked or not -- in a variable? Then you can > manage individual number blocking in a much cleaner, more extensible fashion. > > Feel free to ignore me if it sounds like I'm suggesting you walk all the way > to