similar to: Looking for recommendations - US SIP provider for T.38 Faxing

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

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.

Hi there, I have been asked by a industry group of University educators & leaders to present a session to them at an upcoming conference on Open Source philosophies. I've pretty much completed my presentation outline and notes, and I'd like to include any case studies regarding the benefits (both financial and flexibility) that any medium to large organization may have

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

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

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 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

I have an Asterisk 1.4.2 system online and have built up quite a large blacklist of tele-spammers that have been calling us. Recently we swapped one of our DID numbers to a SIP provider that now prefixes all calls with +1xxxxxxxxxxxx in front of US numbers (we're in the USA) and I need to edit my blacklist database so that all numbers in there that don't have the +1 prefix in front

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'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

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

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 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 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

I'm trying to use func_odbc to write to a MS SQL db. Here's my func_odbc conf: [OPTIN] dsn=MSSQL-Optin write=INSERT into OptIn (orgID) values (${VAL1}) Dial Plan exten => +18665551212,n,Set(ODBC_OPTIN()=dave) When I do an odbc show, it shows that I am connected to the db. If I use isql, I can write to the db, however, when I use func_odbc, a record will not write. I'm using

I have a small Asterisk 1.4.2 system that I run out of my home based business, and my Dialplan has it set to send any incoming call to my desk Grandstream phone. Works really well. When I leave the office, I need to re-direct the calls to my cell phone. I tried to do this through my Grandstream phone, but it really didn't work all that well. Callers would tell me that they tried to

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 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

I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there.  I ran into an issue when testing that incoming calls from MagicJack would go silent after about 10 seconds.  This happened while in the automated attendant area.  This problem did not occur with Asterisk 13 LTS.  I reverted PJSIP