similar to: Douglas Garstang <dgarstang@oneeighty.com>

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.

Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI> show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2

It seems that I am having a heck of a time explaining my attempts at distributing ACD Queues to the list. Here's one little problem, that's only a piece of the puzzle. dundi.conf: 180q => global_dundi_q_pbx1,100,IAX,dundi1:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => global_dundi_q_pbx2,200,IAX,dundi2:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q =>

I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a

I'm having some issues with Background() and Read() commands. See the example below. This is when I wait for Background to finish playing the sound file, before entering '12345#'. All works fine. hestia*CLI> -- Executing Answer("SIP/2944093-3366", "") in new stack -- Executing Wait("SIP/2944093-3366", "1") in new stack --

Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch

I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in extensions.conf changed recently? exten => 5555,1,NoOp(${CALLERID}) hestia*CLI> -- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel" <2944093>") in new stack == Auto fallthrough, channel 'SIP/2944093-d24d' status is 'UNKNOWN' This

Ok, this is highly confusing. hestia*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 2944030 2944030 oneeighty_start No RFC3581 2944035 2944035 oneeighty_start No RFC3581 sip users (type=friend) are in sip.conf. I have nat=no

Hi All, I use Ubuntu server 10.04 LTS as virtualization platform. Actually running kernel 2.6.32-31-server #61-Ubuntu S root at jupiter:~# uname -a Linux jupiter 2.6.32-31-server #61-Ubuntu SMP Fri Apr 8 19:44:42 UTC 2011 x86_64 GNU/Linux We have on running virtual root at jupiter:~# virsh list --all Id Name State ---------------------------------- 1 kvmtik.4safety.cz

Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and

Did the following: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel svn checkout http://svn.digium.com/svn/libpri/trunk libpri Compiled and installed zaptel, libpri, asterisk and finally asterisk-addons. Following errors ocurrs when compiling

Try to ping from client to server with its hostname. Sounds like dns problem. ping server Then try to ping its ip address. Then try to add server address to host file. Ex 192.168.8.30 server.example.com server Best M On May 26, 2016 12:02, "Nico Speelman" <nico at speelmanrobben.nl> wrote: > Hello, > > I've been trying to add a new server to my Samba 4 Active

Does anyone know if realtime extensions allows extensions in the format callerid/extension yet? ie the extensions.conf file allows you to do: 5551212/1000 => exten ... and it matches against extension 1000 when the caller id is 5551212. Last time I checked, realtime didn't support this yet. That's a show stopper for us. -------------- next part -------------- An HTML

Hello all, I need to write a queue_log parser that is going to implement more or less the functionalities described here http://lists.digium.com/pipermail/asterisk-users/2003-July/014965.html of course not everything from scratch, but this is where I'd like it to go. I am looking for - previous work (maybe it's ready somewhere and I've never heard of it) - suggestions -

All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk.

Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug.

Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? Doug

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says "the context for the voicemail box that we're looking for in the dialplan for the jump to the

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);