similar to: Help setting up IMAP_STORAGE on CentOS 5

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'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'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 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'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 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

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 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 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

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 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

The easiest way is to just run the Dial() command to forward the call to the hard fax without ever Answer()-ing the call. Without an Answer() on the call, Asterisk can't listen for fax detection (because the call hasn't been set up and there is no audio leg yet). Thank you, Noah Engelberth -----Original Message----- From: asterisk-users-bounces at lists.digium.com

Hello, our gateway receive incoming calls from an outside gateway for multiple DIDs. For some of them we want fax detection, for other no. To do so, faxdetect is set to yes, but how to disable the fax detection for a specific incoming DID? For those DIDs, we just want to forward the call to a real fax machine DID which will do the job. Thanks for any hint Regards -- Daniel

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

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'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

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

Hi all, I've got a provisioning file that I use to configure Cisco SPA112's. I'm wanting to get this file to do 3 things for me. I want to turn T.38 on, Call forwarding off, and Call waiting, off for both lines. but it's not working. This is what I'm using to enable T.38 for line 1. <FAX_Enable_T38_1_>Yes</FAX_Enable_T38_1_>

My system is Debian I have successfully installed Asterisk 1.6.1.6 to a non standart folder /opt/asterisk Then for asterisk-addons-1.6.1.1 I run: ./configure --prefix=/opt/asterisk/ make menuselect And then command "make" fails with following errors: Generating embedded module rules ... make[1]: Entering directory `/usr/src/asterisk-addons-1.6.1.1' make[1]: Leaving directory

Hopefully someone can point me in the correct direction. I had a 1.4x system die on me yesterday, while I was prepping a new machine to replace it. Took the machine on site yesterday and spent the day and part of the evening getting things working. This morning, I finished up converting my dial plan, knowing there'd be calls of things that I missed. While testing, I've noted that all