similar to: Termination for Asterisk Users - Inter-Asterisk Exchange

I expropriated the right to rip Daniel's disclamer for use in my email too.. DISCLAIMER: You might have to pay royalty fees to the G.729 patent holders for using their algorithm. For easier testing I prepared codec_g729.so binaries and associated libraries and put them on the web: http://kvin.lv/pub/Linux/Asterisk/ arkadi.

Good day all I've tried my iax conf and I'm struggling.So I want to know If someone else got this working and if they can pleas send my their configs I have to asterisk server,in different tows,both offices connected wit a direct line so both servers are on the same network running SIP.Each town got different extension register to each sever.Town A=100+ town B=200+ How do I get town A

I purchased an IAX2 hardphone, X100 otherwise known as a Netweb X100 or YWH100 with a PA168 chip and the latest firmware 1.45 available, from a US retailer. I was able to configure the phone to work with my Asterisk box, except the hold and transfer buttons do not work. When you press the hold button, it rings endlessly, the transfer button, displays "transferring" but it does nothing.

For those of you who may be interest.... IAX2 loads are now available for the standard builds... http://www.aredfox.com/edownloadsiax2.htm Just a word of caution... Remember to change the ports over to 4569 from whatever. And don't forget to grab the palmtool from http://www.aredfox.com/download/tools/PalmTool.zip My own testing of IAX2 with both a phone and an ATA is that IAX2 is

Folks! This is to let all of you know that I am making D'Link make an all out effort to make D'Link Phone DPH80 and DPH100 work with Asterisk. I have provided the Asterisk Platform to D'Link's R&D Division located in Goa, India, where their IP phone's SIP Bios is undergoing modifications based on my recommendations/suggestions. I have also provided the test bed &

I'm just wondering if anyone knows the story with these... http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5721202362&ssPageName=STRK:MEWA:IT He claims they support IAX2 and SIP... but almost no history on the account selling them. I didn't see anything in the wiki about this company either.. Does anyone have any history with these phones? Thanks, Jared

Greetings, I just received some netweb-301 phones frm Seshu down in NJ. I cannot for the life of me get it to register with the asterisk server, nor upgrade the firmware to the latest (1.41) i'm still using 1.37. The packets are traversing the router, going into the other subnet, hitting the asterisk box, but not actually making it to asterisk. Nothing in the asterisk logs, but tcpdump

Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Most of the VoIP phones I've looked at only have 4-6 line presentations; is anyone aware of one that has more? I tried to get some info about Snom's Keypad 220 since it has loads of programmable

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I'm running asterisk stable 1.0.5 and I'm trying to get the netweb eezee phone version v1.37.008 to talk IAX to asterisk. The pages I saw in the wiki maybe didn't hold my hand quite enough and the information on the eezee phone website appears to be for a different firmware version. If anyone has done this recently and has a working situation I would appreciate some useful hints

List, The D'Link phones are not reliable at this time. I am trying to get them fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk. Seshu Kanuri "G

There was some discussion in the past about which one is the best Content Management System that can be used in conjunction with Asterisk. Mambo was supposed to be the best out there under GPL. The guys who developed Mambo have a new product now - Joomla. I am using this and it appears to be better than Mambo in many respects. Read the gist about Joomla below. ------------- If you've read

I am trying to download the latest release of Asterisk from: ftp://ftp.digium.com/pub/asterisk/ The link provided by Digium is incorrect for the Asterisk Tarball as there is no such file at ftp://ftp.digium.com/pub/asterisk/asterisk-1.0.7.tar.gz However the links for the Asterisk-Addons and other Tarballs is OK ftp://ftp.digium.com/pub/asterisk/asterisk/asterisk-addons-1.0.7.tar.gz Does anyone

Folks! I dont know whether anyone has done this exercise before of putting together a Wish-List of things that you want to do, if you have all the gadgets you need and have a client base that needs Asterisk's Features and more. Here are some of the scenarios I am playing out that I will do, once I have enough time. Can anyone add to this list of Scenarios with or without using other gadgets,

I have setup Grandstream to connect to my Asterisk Server. All the digits 0-9 are accepting dtmf. But When I try to send the call by Pressing # Key, nothing happens. Does anyone has a standard configuration for Asterisk and Grandstream as a PDF file or something to see. How do you send the connect signal? Seshu Kanuri -----Original Message----- From: asterisk-users-admin@lists.digium.com

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI

Why is no one suggesting any solution here for this problem, which has been lingering for a while. Are we not supporting H.323 on Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of ruixun wu Sent: Thursday, July 22, 2004 4:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] still can't

Folks! Let me clarify something to the Asterisk community about the CDR tool. 1) This is *not* my code to start with. I picked the original code from this forum here... http://www.voip-info.org/tiki-index.php?page=Asterisk%20CDR%20Areski%20GUI 2) The original code was not working (for most part, as the MySql portion has bugs) and I fixed this and added a few bells and whistles. 3) The

I have a couple of Polycom phones, bootrom 2.5.0, SIP 1.3.1.0056. Works great with Asterisk when I power on the phone. However, after some time, say an hour, I cannot receive calls on this phone. On Asterisk, when I do "database show", it does show the phone in there, but it cannot reach the phone. Is there a way to keep alive the connection with Polycom? Any suggestions? Thanks, --

I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux on this router and included a bunch of Linux tools, one of which is Bandwidth Management. The QoS aspect of this is supposed to be much more granular than the previous solution (Wonder Shaper). I have not been able to find any suggestions for how to impliment QoS (Bandwidth Management) using the web interface of Alchemy.