similar to: Help! how to deal with relational databases

Hi all, My latest Asterisk system is based on Debian squeeze with Asterisk 1.6.2.6-1 and SIP only. One of my favorite features that I had working with Asterisk 1.4 is the PrivacyManager. However, this was not straightforward, because anonymous SIP calls arrive with ${CALLERID(num)} = "anonymous", instead of being blank. So, to get it to work I added the first three rules to

Hi, Because I want to create several draggable elements in a list, I''d like to do the following: <script type="text/javascript"> <%- elements.each do |element| -%> <%= draggable_element_js("element_#{element.id}".to_sym, :revert => true) %> <%- end -%> </script> However, if you''d try this in your app, you''ll

Hi all, Previously, I had the PrivacyManager working for me exactly as would be expected, but after upgrading the OS to Debian lenny and Asterisk to v1.4.21.2 that's no longer the case. Anonymous callers are still confronted with the PrivacyManager, but now no matter what I set the minlength value to, e.g.: exten => jaap,n,PrivacyManager(1,1) ... (I'm not using a

Hello all, I tried to include a file from within the accounting config-file. The filename was specified using a parameter in params as: ACCFILE=/var/lib/shorewall/accounting.generated and then included in accounting as: INCLUDE $ACCFILE However when (re)starting shorewall, it gave some error about being unable to find ''/etc/shorewall/$ACCFILE'' (with the $ACCFILE parameter

Hi list, After wrestling with the voicemail system for a while (Asterisk 1.4.14, Debian etch), I got it to work, but I still have lots of questions, like: * Why can't I delete any voicemail messages? (Response: "Message undeleted.") * Why can't I listen to the messages in the Old folder? * Why can't I use the advanced options? (Response:

Hi folks, My client and I are having a problem getting a portable Esaote ultrasound machine to connect to a Samba server. The unit has an integrated laptop with a Windows XP version that can hardly be modified. Upon delivery the vendor only changed the user name and workgroup for us. When I asked for the user password to make a matching Samba account, the vendor refused because they use

https://bugzilla.mindrot.org/show_bug.cgi?id=2738 Bug ID: 2738 Summary: UpdateHostKeys does not check keys in secondary known_hosts files Product: Portable OpenSSH Version: 7.4p1 Hardware: amd64 OS: Linux Status: NEW Severity: minor Priority: P5 Component: ssh

Hi list, Hopefully, some of our Dutch members can help with this one. I'm also based in the Netherlands and am using a Sipura (Linksys) SPA-3000 (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test system. It works fine, except that the Called ID (CID) is not working. I'm aware that KPN (our local telco) requires a separate subscription to activate CID on POTS

Hi folks, My site uses puppetmaster 3.2.3 together with Debian wheezy, but I want a few packages from wheezy-backports, like iceweasel and libreoffice. The former I''ve installed as follows: package { "iceweasel": ensure => ''22.0-1~bpo70+1'', provider => ''apt'', install_options

> I'm curious which version of the source are you working with? I was working on 2.0... :-) > I haven't sync'd to the SVN trunk in the last day and half. I'll do so > now and check. Don't bother then, I should have synced to CVS and made sure those fixes didn't already exist. After I posted my previous email, I found two more problems. One is a function

Hi list, After a recent upgrade to Asterisk v1.4.14, my message log is now filling up with the following error messages: <-------------> [Dec 29 17:24:52] WARNING[10655]: chan_sip.c:6645 determine_firstline_parts: Bad request protocol Packet --- (1 headers 0 lines) --- bitis*CLI> <--- SIP read from 82.101.62.99:5060 ---> Cirpack KeepAlive Packet <-------------> Seeing

Hi folks, Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1. As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version can support IPv6. However, it seems that I can't get it to support both IPv4 and IPv6 at the same time. For example, if in sip.conf I set the bindaddr variable to '::' it will only listen on IPv6 and none of my IPv4-only

Hi folks, Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9 to 1.8.13, my server is no longer able to register a connection to a SIP account at my ISP (XS4ALL in the Netherlands). At the same time, it is still able to register a different account with another SIP provider, so it must be that they no longer have the same basic requirements. The relevant part of my

Hi all, For a while now I've been using Asterisk together with HFC-PCI cards (Cologne chipset) for Euro-ISDN BRI support. However, I do not consider this to be the most reliable solution and believe that the most stubborn problems have always been software related. If my clients are willing to spend a bit more money on different hardware, what do you think the best solution would be?

Hi all, On an Asterisk/Zaptel 1.4 system, one way to gather diagnostic info is: ~# cat /proc/zaptel/* Span 1: ZTHFC1 "HFC-S PCI A Zaptel Driver card 0 [TE]" (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 "HFC-S PCI A Zaptel Driver card 1 [TE]" AMI/CCS 4 ZTHFC2/0/1 Clear 5 ZTHFC2/0/2

Hi list, The default file name format for touch monitor (automon) recordings is: auto-${EPOCH}-caller-calee It's possible to use the ${TOUCH_MONITOR} variable to change the 'caller-calee' part, but what about the 'auto-${EPOCH}-' part? I've been trying to use ${MONITOR_EXEC_ARGS} to add some more commands after the somix sequence for mp3 conversion. This should

Hi all, Perhaps someone here could help me with this. I'm new to Asterisk, but have already met with some success at getting my first system to work with two different VoIP (SIP) providers: XS4ALL and InternetCalls.com. The config for the former works fine, but my InternetCalls.com config works only intermittently for incoming calls. It currently looks like this: [general] port=5060

Hi folks, Recently I learned how to configure libvirt with USB pass-though functionality. In my case I configured my guest domain with this block of code: <hostdev mode='subsystem' type='usb' managed='yes'> <source> <vendor id='0x0c93'/> <product id='0x1772'/> <address bus='1'

Hi list, My Asterisk v1.4 system now has two ISDN channels and two SIP channels. The idea is to make a dialplan that mostly uses the SIP channels for outgoing calls, but I'd like those to fall back automatically to ISDN if the SIP channels aren't available, possibly in combination with a warning issued to the caller before the call is actually placed. Is this possible with

Hi list, After recently setting up voicemail for Asterisk 1.4.14 on my Debian etch server, I noticed that I can't delete any old voicemail messages. The voicemail menu option "Press 7 to delete this message" is available, but when I press 7 the response is always "message undeleted" and the message is still there. What could I be missing here? Thanks, Jaap