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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 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 =>

We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. So far, this has happened on both times the

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

I've been spending the day trying to get IMAP_STORAGE on my test box, to evaluate for production, but I'm having no luck getting uw-imap to build. I've tried installing it from an upstream package, but Asterisk still isn't finding it to compile -with-imap. My google searches have turned up very little for documentation on dependencies, gotchas, etc for either item, so I'm

Hi there I am new to rsync so be gentle Ok i install cwRsync and i used the script that came with it but just added my filename and remote computer. right now it says rsync -r -delete /c/backup/ 192.168.1.152::/c/backup i also tried it with rsync -r -delete /cygdrive/c/backup/ 192.168.1.152::/c/backup and rsync -r -delete /cygdrive/c/backup/ 192.168.1.152::/cygdrive/c/backup the error

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'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

Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short

I currently run an Asterisk 10 system with hotdesking functionality set up. Several of the users have worked with a system in the past that supported BLF on their IP phones, and would like their current phones to behave in a similar fashion. Right now I have a really kludgy system that mostly works, but doesn't consistently trigger the cleanup macro to "clear" the device state on

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

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