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Hi, Long story short - I have an ancient Britsh Telecom phone attached to my Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call quality is excellent. However, dialling out is impossible, as Asterisk consistently mis-reads the number of pulses the dial sends (it could be a squiffy dial, I'm not sure). Not to mention the fact that, in today's modern "want

I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse & repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 00000

Hi folks, I'm running Asterisk 11 (at the moment - planning to u/grade to v13.7 LTS), I've just configured the voicemail function, and it's mostly working fine... except when I try to leave a voicemail! This crashes asterisk with no entries in the messages log. The system is running on Centos 6 (or maybe 6.5, I'm not sure how to check this). uname -a returns: Linux

On Fri, Jun 19, 2015 at 2:14 PM, asterisk <asterisk at solutionengineers.com> wrote: > > Hi, > > Long story short - I have an ancient Britsh Telecom phone attached to my > Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call > quality is excellent. However, dialling out is impossible, as Asterisk > consistently mis-reads t...

Does such a beastie exist? I've tried a couple of UT Starcom WiFi SIP phones (the F1000g and F3000 respectively), and found them both to be seriously lacking - regular crashes (especially the F3000), poor battery life, and poor reception in particular. However, whilst SIP phones are great, I'd really like an IAX2 phone if there is one, as I can make that work "natively" though

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

Hi... I have what is, I am sure, a relatively common & straightforward problem (no, NOT that kind of problem!)... I'm trying to hook two asterisk servers together so I can make a "distributed" PBX. Here's the scenario: [MASTER] is in the office. It has unrestricted access to the internet, and a fixed IP address. It has 3 SIP hardphones configured & working, plus a

I have 2 asterisk servers - serverA and serverC - connected via IAX2. On serverA, I have a "telemarketer hold" extension which, if I transfer a caller into it, loops around playing music & "please wait" messages, until they give up & hang up the phone. Also on serverA, I have a custom devstate, which lights a lamp on a phone connected to serverA, which tells me if

Hi, I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there are 3 BT lines connected directly to these ports. One of the lines has BT FeatureLine Compact and this is the line I am having problems with, the other 2 lines are working perfectly, detecting CID, answering incoming calls and placing external calls via SIP devices. I am receiving a error log entry: chan_zap.c:

...<op.txci53z43wzjep at pc-lenz> > Content-Type: text/plain; format=flowed; delsp=yes; > charset=iso-8859-15 > > > You may want to start from here: http://astrecipes.net/index.php?n=204 > l. > > > On Sun, 19 Aug 2007 00:46:45 +0200, Ade Vickers > <javickers at solutionengineers.com> wrote: > > > Hi... > > > > I have what is, I am sure, a relatively common & straightforward problem > > (no, NOT that kind of problem!)... I'm trying to hook two asterisk > > servers > > together so I can make a "distributed" PBX. &gt...

I recently implemented a simple "spam trap" extension for telemarketers - once identified as a telemarketer (usually they ask to speak to the person in charge of recruiting/website/purchasing/etc.), I simply offer to put them through to the person in question, & dump them on a special extension which plays music for 15 seconds, then 1.5s silence, then a "please wait, we're

Hi folks, I've been playing around with an Asterisk server in my office for a few weeks now, and I've got it pretty much nailed down the way I want it, which is nice. One of the features I'm using is the ability to switch different contexts in & out of the dialplan on a schedule. So, for example, I've got the "official" tel number ringing my desk phone between