similar to: Looking for PSTN provider with unlimited inbound/outbound plan

Hi all, I've searched the depths of the resources on the internet; however I'm having trouble deploying a pxeboot solution via pxelinux. So far what I have successfully implemented in my infrastructure is a successful pxeboot setup for FreeBSD ( without the use of pxelinux ). I'd ultimately like to have a solution that will allow me to choose a network install of various Unix-like

Hi all, I'm looking for some rudimentary insight on GotoIf() which seems to be failing on me in my dial plan. All I basically wish to do is block a particular caller. Sounds easy enough, but my ternary operator/plan currently is not properly being implemented. Can anyone spot where I'm being a momo? All extensions get forwarded to the following macro: [macro-forward] ; arg1 = phone

Afternoon all, Is there any particular reason why yum fetches rpms for two architectures on almost any update/install I'd like to perform? This includes both i386 and x86_64. Here's a small excerpt after performing a 'yum update' cups-libs i386 1:1.1.22-0.rc1.9.18 update 107 k cups-libs x86_64 1:1.1.22-0.rc1.9.18 update 112 k

Hi all, There's a myriad of options these days and I haven't been keeping up to date with what's respectable any longer. I essentially need a provider that will provide me with one DID to start and let me trunk over SIP or IAX2. I'd like to obviously trunk with Asterisk on my end and have full control over the dial plan. This way I can branch out my DID into extensions and have

Anybody have a recommended provider which supports TLS for SIP trunk communications, or even encryption via IAX? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140421/2e46203e/attachment.html>

Hi all, I have the following in my extensions.conf: exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" | "8585970327"]?15:5) The numbers listed above are known spammer numbers. However, when I call from any other CALLERID, it still directs me to s,15 which is the Hangup() application. Here are logs from the asterisk CLI: -- Executing

All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. Any suggestions? - sf

Hi all, I've compiled zaptel drivers and reconfigure asterisk afterwards from source --with-zaptel. Modules are loaded accordingly: asterisk-1.4.2 # lsmod |grep z Module Size Used by ztdummy 5472 0 zaptel 194504 5 ztdummy crc_ccitt 3521 1 zaptel my musiconhold.conf: asterisk-1.4.2 # grep -v '^;'

Hi I am wanting to install : dovecot 1.0-stable I currently have dovecot 0.99.11-1 which came packaged on Fedora Core 3 It is doing auth off a MySQL db an working well I have downloaded the latest stable tarball and extracted ... I don't see how to enable mysql lookups at compile time... Help please ... I am getting a lot of pressure from the people at work ... Regards Andrew Andrew

I know that Rails 3 ideal of unobtrusive javascript will result in the removal of all inline javascript. But where does that leave RJS? It looks like PrototypeGenerator is still a part of the Rails core, so I''m going to assume that RJS templates will remain unchanged. Still looks like gems/plugins such JRals will be necessary for use of other JS frameworks. Can anyone confirm or deny

I use Trixbox 2.0 with a Sangoma A200 I also have echo so bought the HPEC and yes it works brilliantly. The problem I have is I used to use Trixbox 1.? with this sam hardware and had a few inbound CLID issues on my UK BT lines, Sangoma support suggested changing the RXGAIN in zapata.conf and it worked. Now using Trixbox 2.0 and have upgraged to latest software Asterisk 1.2.16 Zaptel 1.2.15

Greetings, I was hoping to get some feedback on a plugin I wrote. The plugin is called conflict_warnings and is currently available from my github repository at http://github.com/EmFi/conflict_warnings The purpose of the plugin is to provide a simple methods of preventing data inconsistencies that could arise from multiple users interacting with the same resource. Under basic operation a

My apologies for the off-topic post ... No matter what settings I try, when I dial in to the SPA-3000 on the PSTN line, it picks up the call and immediately gives me a fast busy tone then hangs up. The info tab says under PSTN Line status: Last PSTN Disconnect Reason: PSTN Disconnect Tone which seems to indicate that the SPA thinks the caller has hung up. Since I am in Japan, it is possible

Hi, I am using the following script to limit my outbound traffic. This scipt runs on a box behind my firewall. It limits my outbound passive ftp traffic to 39K perfectly....just like i want. However, i just noticed that it is also limiting uploads coming to my server. Is there something I can change to make it not limit uploads to my server? #!/bin/bash #shaping passive ftp traffic #

Greetings list- I have a box with a single FXO card in it. I'm able to dial out ZAP/1 with no problems and as expected. However, I would like inbound calls on that POTS line to go unanswered by Asterisk since I have other equipment on the line. I've setup zapata.conf for the channel without a context but the line is still answered. I've also setup a blank context with the same result.

Is there a way to specify the outbound leg Call-ID? -- Eric Chamberlain

I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is

Hello I get this warning all the time when I am using iax2 for inbound calls or outbound. Aug 24 13:48:41 WARNING[-1105474640]: chan_iax2.c:4873 socket_read: Error: Resource temporarily unavailable I get the calls and the sound is fine. But the screen on the cli is full of these warnings and Error: What can I do to fix this. I get it when using calls to iaxtel, FWD, VoicePulse, Nufone and

Hello All, I would like to record inbound and outbound calls to and from one number. I tried to add lines to my extensions.conf: DAY=`date "+%m-%d-%y_%H:%m"` ;outbound exten => 5555551212,1,Record(${DAY}:gsm) exten => 5555551212,2,Dial(${TRUNKL3}/${EXTEN}) ;Inbound [line2] exten => 5555551212,1,Record(${DAY}:gsm) exten => 5555551212,2,Dial(SIP/101,20) exten =>

silly me, I thought the inbound would be the hard part. how little I knew... can someone please give me any insight into why outbound is not working, in fact why trying to enable outbound fouls up everything? I'm using asterisk, most recent from cvs, I'm behind a nat, and I'm trying to use iconnecthere.com for outbound and inbound. Inbound is working fine, no problems. But for