similar to: Strange Asterisk/SIP call forwarding behavior

I have a working dialplan for our phone system with Mon-Fri, business hours identification, etc. But what I'm lacking right now is support for company holiday dates. What I'd like to do is to create a database of these dates and just update them as new years rollover. I suspect others have done this sort of thing with Asterisk before, but I've not found any resources so far.

I have a new Asterisk system going into production next week and I'm a bit stumped as to the best way to handle the Dialplans for it. The Asterisk system is replacing 4 separate PSTN lines with both SIP & PSTN inputs. The setting up of the dial plan is giving me some design headaches, which probably means I'm missing something obvious and doing this the hard way. I have separate

Hi all, I'm trying to get my head around our Asterisk network configuration. We've been using it for about 2 years now (home office) and it works great. Its Asterisk 1.4.2 with SIP through external provider(s). We have the Asterisk server behind our IPCop firewall, and have a dedicated IP address that comes to the firewall from our ISP (Cox) and that is routed to our Asterisk box

I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming & FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its

For some reason our Asterisk box is doing something really unusual following applying a routine update to CentOS 5 on Monday. We have Asterisk 1.4.2 and its been working great for years. But now when the phone system receives an incoming SIP call, its not providing any audible dial sound to any caller. It is recognizing the incoming call, and after no answer for about 5 rings or so, it goes to

I have an Asterisk system setup for our small business, and its working well. I posted to the list about a week or so ago, regarding having it handle direct extension dialing, and unfortunately I'm not any closer to solving this issue, so I was hoping someone might have a working example of how to set this up they could point me towards. Basically I have everything EXCEPT direct

I am revising our DialPlan strategy for our Asterisk system (1.4.2) and looking for some info on 'best practices' for this. Here's what I'm trying to do: I have an ACD menu that gives the caller the options as follows: - Press 1 for sales - Press 2 for support - Press 3 for customer service - Press 8 for a 'Dial by Name' list or enter the extension number at anytime

We have added Asterisk to a line of 'mission critical' servers at our business, and being in the web application development business one of the core things we do is to monitor web server availability. I'd like to add Asterisk to the servers that our monitoring systems are handling, and also that our SIP trunk provider has our Asterisk system correctly registered at all times.

I am not finding anything relating to this on Google, so I thought I'd pose the question here... I am running Asterisk 1.4 on a CentOS 5 Linux box. I needed to use a custom built PHP5.2.10 install to interconnect with our Firebird SQL database, which I've done. But I noticed that the default install path for PHP5 on this box appears to be /usr/local/bin/php rather than the path

I have an Asterisk 1.4.2 system installed at our office, and have a few 'on the road' sales people that want to make calls from their cell phones in transit, but they are complaining that people returning calls that they make from their cell phones are simply just using the CID that is coming from the cell phone which is causing them to get phone calls outside of business hours. What

I am trying to implement a change to our Dialplan that will thwart tele-spammers that are calling us with blanked out caller ID. The caller IDs seem to vary between originating callers when they block caller ID. I've seen the following: "anonymous" "" So I'm checking for these. However recently one company seems to be bypassing this, so what I wanted to do was

I am running Asterisk 1.4.2 and recently we changed the SIP provider of our main incoming DID number. The new provider prefixes all CallerID records with a +1 in front of the number, whereas the previous SIP provider did not. Consequently now all my blacklisted numbers aren't matching in my Dialplan, so I'm getting tele-spammed. Is there a way that I can work with the blacklist

Is there a simple command in the CLI or other for Asterisk 1.4.2 where I can list all numbers in the blacklist database? I need to export this data to another database, but am unsure how to get to it all in a list. Thanks in advance for any pointers. Myles -- ----------------------------- Myles Wakeham Director of Engineering Tech Solutions USA, Inc. www.techsolusa.com Phone +1-480-451-7440

I'm pretty new to this whole Asterisk system & VoIP thing, but being a programmer by trade the complexity didn't scare me off (at least not yet)... I have setup an Asterisk system for my home & home office. My wife & I run two separate businesses from home, and we have a general family home phone line as well. The cost of all these lines with analog carriers was getting

I have an Asterisk 1.4.2 server with 3 different SIP providers and Asterisk for Skype gateway installed. Periodically the SIP providers go offline for some reason, or the Skype connection fails. When this happens, I lose my SIP registration to the provider. Unfortunately I don't know this has happened until someone eventually contacts me to say, "I tried to call you but it

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm