similar to: CALLERIDNUM not passed over?

I threw together a web gui for the static database configuration over the last couple of days. I built it using mod perl and the template toolkit. If enough people show an interest in this I'll put up a distribution, although it could take a few days. The interface is as generic as possible so you can throw pretty much any asterisk .conf file in and it works. The interface assumes you

Hello, I have a queue with only one element and one agent member. I want that my call leave the queue after 30s. My problem is that my call stays 60s in the queue and my agent is called 2 times. Can you say me how can i do it please?? -------------------------------- [queue] music=default strategy=ringall timeout=30 maxlen=1 context=mbdsys announce-frequency=0 announce-holdtime=no

Hi Does anybody know how to build the LibIAX2 from Virbiage? It has some nice features when using Firefly (Messaging, Status Indication). The source can be downloaded here: http://www.virbiage.com/3rdparty/. It does not contain any directions how to compile. Any hints? Thanks! Reto

try this one exten => 999,1,Answer() exten => 999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To:

I have no idea what this really should be called, so for lack of a better name, I called it "reverse hold". Hopefully someone else can make use of it, or even make it better, as its the first thing of its kind I've made for asterisk. Like most people, I'm very busy, so when I call other companies, sitting on hold really sucks. If you have speaker phone, its not so bad, but then

Hi All trying to compile asterisk under linux kernel 2.6.6. Currently under zaptel get the following error make linux26 Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 as going from the readme. is 2.6 not compatiable with asterisk and should I go back to 2.4.26. Also has anyone got the sipura 3000 working with asterisk, both fxo and the fxs ports on the

Hello, I have modified the VoiceMailMain application to satisfy the request of the "local users", i.e., my wife. She lost patience with too many options (we have one mailbox, so we don't need to forward messages, or reply to messages, or file them in 6 different folders...) So the modified app says "Message 1", reads the message, "Message 2", reads...

I've just compilied th latest CVS of * with USE_MYSQL_FRIENDS enabled ("1"). During startup * tells me that it connects to the db, so this should be fine. Nevertheless I don't see any users from the db when I run "sip show users" or "iax2 show users" although I configured some. It is also not possible to call them. Any hints?

I've done something similar at home, but made my dialplan such that I can dial either 10 or 11 digits locally. I don't use a "throw away" digit at all. Any 7, 10, or 11 digit call will be appropriately mangled and sent out the PSTN / VoIP provider. ________________________________ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On

Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). ?We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. ?Attempts to send outbound calls on any Zap channel will result in hearing a loud

I have a brand-spanking new TDM31B (3 FXS, 1 FXO) and when I start wcfxs (the only module that recognizes the card) from Zaptel 0.9.1 I get: Zapata Telephony Interface Registered on major 196 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO FXS Module 1: Installed -- AUTO FXS Module 2: Installed -- AUTO FXS

I've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack

Hi all, we use OpenSER together with Asterisk. All SIP users registers with OpenSER and asterisk is doing the voicemail thing. We use the Asterisk RealtimeArchitecture for voicemail users and SIP peers. The database table for the sip peers is a view from the OpenSER subscriber table. The MWI for a user will only work, if the user object (sip peer) is loaded into memory and visible with the CLI

Hi there, I was wondering how I can use setgroup and checkgroup for perfom incoming and outgoing limitation checks. I've have some users that doesn't what to be able to recieve more than 1 call at a time, and I also want to limit a users outgoing call abilities. Any help would be greatly appreciated. Kind regards Cf --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus

Hello everybody For network simulation purposes I am trying to run a Linux image with a PAX enabled grsec kernel on a Gentoo xen-3.1.1 with HVM. While the image boots flawlessly on real hardware the kernel does not really like the fully virtualized Xen/Qemu environment. It does not succeed to boot (for dmesg see attachment). I first tried with the grsec- patched 2.6.14.6 sources but it

Is there a way I can have "voice mail check" calls coming from my internal users automatically get to the right extension, without having the user enter their extension? I'm thinking that I could have the local SPA boxes translate, or have each user live in a context where the extension in question exists uniquely per user, but both of these seem kludgey. Thanks in advance for

Hi: I'm attempting to run "The Neverhood" on FreeBSD/wine-2002.10.07 no windows config, which, according to the application DB, should work. When I run the setup95.exe program the Neverhood installer screen, and when I hit "INSTALL" I get: bash-2.05b$ wine d:setup95.exe fixme:cdrom:CDROM_GetIdeInterface not implemented for BSD fixme:reg:GetSystemInfo not yet supported on

Hi! I've read in the asterisk docs (AstARA.html) that realtime users/peers can't be notified (MWI with SIP NOTIFY) when they have new voicemail messages, because their object are not persistent in memory?! But that's what we really need!!! Is their any work in progress to get these things working? Or are their any known workarounds? I'm using asterisk version 1.2.1 with OBDC

Hi, For some reason, I seem to have issues with dailing toll free numbers and can't seem to find out why, sometimes, I get a busy signal. Some other times I get weird errors from the phone. The error below was a simple busy signal. Here's couple of my info relevant to the problem: -- Reconfigured channel 1, PRI Signalling signalling -- Reconfigured channel 2, PRI Signalling

Hi there Is there an easy function available in R for calculating the running mean? Reto -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or "[un]subscribe" (in the "body", not the subject !) To: r-help-request at