similar to: CVS File README.mysql concern..

Hi, Does the G.729 module support adding more licences??

Hi, I am still toying with the idea of going ahead with using the G.729.. Can those using it tell me about some of your experiences using G.729.. Things like and problems you had running it, the voice quality and anything else you can think of... I have read in the archives that asterisk has to be run with -c.. Is this still the case? and if so does this mean that * can't be run using the

Hi, Does anyone have any instructions or pointers to get VM2 to work with MySQL? I can't seem to find any docs or how-to's on this.. Later..

Hi, Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs running g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa

Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don't have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and

I have an X100p card....and it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael

Hi All, How to park and pickup a call? The scenario of park and pickup described as below. UserA made a call to UserB, and the call ware connected, Then UserB parked (or hold) the call, and told UserC to pickup the call on one line, and then, UserC pressed some keys to pickup the call. Who can tell me what's the Park/Pickup's typical flow in the Asterisk. And how to set the sip.conf,

hi, I'm currently trying to use Speex with Asterisk from my OpenH.323 client. It seems to mismatch the codecs, below is my log from Asterisk. My Openh323 client crashes in responding to a Speex request for bits per frame. I'm guessing it either isn't running the codec correctly or doesn't support the same subset of speex codecs as openh323. (I'm using speex-1.0.1 with

SQL help needed.. If I have a MySQL table with dialing codes and a corresponding description (see below) and I want to lookup the best match for a phone number.. What would the SQL look like to do it? or would it take more than just SQL to get to the best result? Thanks.. Later.. Example numbers, (random end digits so I don't know who's they are.) 00442085673456 - UK London

Anyone had any experience with an Asterisk server as a VMWare virtual machine?

Here is the scenario... SIP UA's can use either GSM or G.711 ( in that order of preference in the sip.conf ).. Asterisk Server1 is linked to Asterisk Server2 via IAX2 and also supports GSM and G.711 ( also in that order of preference).. 1. If a call comes in from the UA using GSM and then goes out over the IAX2 leg, Will Asterisk simply move the GSM encoded data from the SIP channel to

So, after reading this, I wonder if anyone has 1.4 and MySQL working... Is there a non-standard version I can download? more /usr/src/asterisk-1.4.0/doc/mysql.txt MYSQL LICENSING UPDATE ====================== We were recently contacted by MySQL and informed that the MySQL client libraries are now under GPL license and not LGPL license as before. Since Asterisk does allow exceptions to GPL,

Anyone got any pointers on where to find USB handsets or headsets that can be used as the audio device on a softphone? Later..

I have the following setup.. [extensions] ; all extensions defined here. exten => 1234,.... exten => 1235,.... [dial-out] ; PSTN dialout config ignorepat = 9 exten => _9,.... exten => h,.... [local] ; phone context in sip.conf is here.. include => extensions include => dialout The question is where will the "h" extension be active?? it appears to run for ALL,

Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, _________________________________________________________________ Expand your wine savvy — and get some great new recipes — at MSN Wine.

I have a CentOS 4 server at home where we store all our data, I wanted to get some idea of the amount of traffic going into and out of the server by protocol so I installed IPtraf.. When I run it from an ssh session it looks fine but when I run it directly on the server console its doesn't look right at all.. It doesn't seem to layout correctly.. Is there something I need to do

Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that

Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and

Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector

Hi there I have Asterisk running beautifully on our test server. Over the past few days I have been tearing my hair out trying to compile various versions of asterisk-oh323 on various versions of pwlib and openh323. pwlib is now up to 1.8.3 and openh323 is now 1.15.2 stable. asterisk-oh323 is currently 0.7.1 I have tried these three with many errors. I have tried 0.7.1 with pwlib 1.5.2 and