similar to: Announce: Asterisk virtual configuration

I repackaged everything into one distribution and cleaned up the installation. Should be a lot simpler to install now. Asterisk VConfig is a platform for virtual hosting of end users on a single instance of Asterisk using the realtime database structure. Right now the functionality of the web interface is limited to a direct configuration interface. As soon as Realtime SIP/IAX is done I will

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

<I apologize if this arrives twice... I sent it first from an non-subscribed address, don't know if that'll get to the list or not --JJ> Hi folks, I have an implementation question regarding bridging on a linux box between a catalyst trunk port and a cisco 26something w/802.1q subinterfaces. So right now, there's no vlan trunking going on on the link my bridging firewall sits

Good Afternoon, I have an Asterisk box that is acting like it is passing "core show" before every command I type. For example, if I type sip, I will get "No such command 'sip' (type 'core show help sip' for other possible commands). Any ideas? -- -jayson

re can somebody tell me what am I doing wrong (at this late night hour), because vlan are not getting bridged at all. ifconfig eth0 0.0.0.0 up ifconfig eth1 0.0.0.0 up brctl addbr br0 brctl addif br0 eth0 brctl addif br0 eth1 ifconfig br0 up vconfig add br0 X vconfig add br0 Y X = vlan-ID (example: 222) Y = vlan ID (example: 223) # brctl show bridge name bridge

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

Hi, examining the network traffic with wireshark shows that asterisk does not set any QoS values at all. What do I need to do to make asterisk set QoS values (on Centos 7)? The wiki says to use vconfig to set QoS values[1]. What does the skb-priority need to be set to? How do you use vconfig on interfaces that are not VLAN interfaces? Is it generally impossible to set QoS values on

Hello - I am looking at connecting 7 - 10 locations together using Asterisk and possibly some VoIP gateway appliances. I need to insure best voice quality as these trunks will be used primarily for customer calls. I am considering implementing a full T1 frame relay circuit to each location which can be done for a reasonable cost. DSL and Cable are currently at each location and setup for

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/311c56ec/attachment.htm

Anybody here having trouble receiving email from the list on Gmail? I havn't received anything since Friday July 29.

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

This is mostly a testing/bug fix release. Hopefully by the next version I will have some real documentation up on the site. Since it's primarily a platform rather than an end user system, without documentation it's not nearly as useful as it could be. http://asterisk.ochsnet.com Chris

I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so... I had to modify the 01-devfs.rules Make linux26 Make Make install... Everything appears to compile correctly but it says module not found when doing "modprobe zaptel" Is this a rights issue? Jordan Novak -------------- next part -------------- An HTML attachment was

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Hi All, I''m using xen 3.0 on CentOS 5 with VLAN. #vconfig add eth0 100 #/etc/xen/script/network- bridge start vifnum=1 netdev=eth0 bridge=xenbr0.100 I know that after xen 3.2 the procedure of bridge configuration is changed. What is the correct command to set bridge interface to vlan? Yukio

Hi, Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide? Thanks and hope the community will not reject my curiosity! :) Best Regards, Vallu Sevana Oy -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have written some monitoring and stat collection scripts that use asterisk -rx "command" The script runs once a min and logs data and posts any critical notifications. Everything is working well with this method but we get the -- Remote UNIX connection / disconnect message once a min and we would like to suppress it. Is it possible without reducing the verbose logging level.

Sending works but on receive it keeps failing - reporting back 'training' failure. I am using Asterisk 1.6 with T38. What should I post to the list to assist diagnoses? Michael