similar to: Asterisk 11 queue calls - emulate Dial(b) functionality

I run a hotdesking system based on the example from Asterisk: The Definitive Guide. Calls come into the [hotdesk] context, which verifies the phone has a logged in user and sends the call to users,${EXTEN},1 if there is a user logged in. The [users] context then includes several other contexts for internal/external call handling, as follows: [users] include => internal include =>

I have a system running Asterisk 11.2.1 that has had a couple calls between internal extensions get "stuck open". I didn't catch the verbose log for the first one, since I generally don't verbosely log to file, but the second one shows that the call that got stuck was dialed, but the caller hung up before the called device answered. This server is running a hotdesking

In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF updates to SIP peers that have subscribed to a hint looking at a Custom device if that Custom device state is RINGING or RING_INUSE. All other states seem to be working correctly. The hint section of the dialplan is: [hints] exten => _3XX,hint,Custom:${EXTEN} Console shows the following for core show

Hello, I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. My plan is to use this handler to update my CDRs with values such as Asterish and Tech cause (see function HANGUP_CAUSE). I want to have my custom hangup-handler be run only once and when "the second channel" hangs up. At the moment, I'm issuing a couple of "channel originate Local/1 at mycontext1

We have several types of phones, cisco 7940/7960 aastra 55i/9113i/ grandstream gxp2010 I want to be able to implement hotdesking where an agent will logon to any phone. I got all of that working, without having to reboot phones, but then hit a brick wall. Voicemail. I still want each phone to use the BLF for voicemail indication, and to use the "voicemail button" to dial

Please point me to samples of popping and wiping hangup handlers. I don't need to use the values returned; I just need to clear any handlers before I push a new one. It's not clear at https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers+Specification how to provide vars on the right-hand side. Cheers, David -------------- next part -------------- An HTML attachment was scrubbed...

Dear Asterisk users, We would like to announce that Druid, Open Source Unified Communications project has just made a major release: Druid 2.0. It is out!It has a ton of new features and a highly improved interface. Asterisk stability has also been greatly improved. For more info http://forums.voiceroute.org/showthread.php?t=837 Some of the key features - Improved Web GUI, faster and smoother -

I am using 1.4, and have the above device, and it worked really well with monitoring 18 "hints" aka devices. Now, I've moved us to a hotdesking paradigm where the user is the "extension" not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ?

I'm trying to set up a way that our users can send an XMPP message to Asterisk (unsolicited) to request information, such as voicemail status or the like. No matter what I set for the dialplan, I'm only seeing Asterisk execute the s,1 priority in the context defined in xmpp.conf for incoming messages, and then the "call" hangs up without executing further instructions. Anything

hi. currently i am running the phonebook in astdb with *database put cidname 0123456789 "name_surname"* and i retrive it with *exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})* Now, my system has mysql and i got all my contacts in there in a database is called *asterisk *and a table called *addressbook**. *password of the mysql is *whateverpasswd* how do i

Hi ?? I have been trying for several days get 3 Cisco spa508g phones (firmware 7.5.5) to work with asterisk 11.6 cert1 and sla. I can get the phones to all ring when an incoming call arrives, and I see the slatrunk working. However the blf function does not work. If one extension picks up the call the others do not show the trunk in use.? And as you might expect the hold and outbound dialing does

I've been working on an interactive XMPP interface so users at my office can interact with the timeclock and queues by XMPP (in addition to IVR menu, which has been running just fine for quite a while before the XMPP interface). I'm using sendtodialplan=yes to handling the incoming unsolicited messages, and typically will have at least one point of interaction where Asterisk requests

I've been tasked with finding and implementing a CDR/Queue analyzer to provide information to management about the call center's performance. My Google-fu seems to be returning a lot of things that are more or less abandoned projects. Does anyone have any recommended solutions for a CentOS 6 / Asterisk 10 "vanilla" server? Not opposed to something commercial, provided it

I'm trying to get my Asterisk 11 test box set up with XMPP, having troubles with JabberSend(). My jabber.conf file is as follows: [general] debug=no autoprune=no [testaccount] type=client serverhost=my.jabber.server username=myaccount at my.jabber.server secret=mypassword port=jabberport usetls=yes usesasl=yes xmpp show connections gives the following output from the console:

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a

2013/3/26 Richard Mudgett <rmudgett at digium.com> > > On 03/25/2013 05:17 PM, Olivier wrote: > > > Hello, > > > > > > I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. > > > My plan is to use this handler to update my CDRs with values such > > > as > > > Asterish and Tech cause (see function HANGUP_CAUSE). >

I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP. One of my customer's endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn't have any NAT between it and my Asterisk system. When the customer transfers a call from one handset to a voicemail box, the NV 7100 sends a RE-INVITE to

I'm trying to to to "dial1" if caller id match: but dial plan execute 220,n(dial1) regardless exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?dial1) exten => 220,n(dial1),Dial(${sales_support}&${accounting}&${family},25,m(penguin)w) exten => 220,n, I was under impression that if condition is met it will jump to

I have a pair of Asterisk 11.5.1 servers operating as a load balanced cluster, with DNS SRV records set up to weight them 60/40 relative to each other (both at priority 0). The back-end is MySQL Realtime, and everything works pretty well with the Cisco SPA phones & ATAs that represent the majority of my endpoints. I recently tried to add an iPhone with the Bria softphone application, to

I enabled the logs on the mysql database and ran : realtime load queues name cou0002-test in the mysql log I can see that the proper select statement is being executed: 2018-12-04T16:29:27.253094Z 229 Query SET SESSION TRANSACTION ISOLATION LEVEL READ COMMITTED 2018-12-04T16:29:27.254384Z 229 Prepare SELECT * FROM queues WHERE name = ? 2018-12-04T16:29:27.254902Z 229