similar to: IAX - lan ip phones.

Hi, all: I am a newbie to the asterisk and its architecture. :( After reading some help in the tarball of Asterisk, I am still in the mess. So I want to know where I can find a detailed explanation of the Asterisk which including the Architecture, Install, Configure, usage example document. Maybe what I want is too much, after all it is a open project, not commercial product. If I want to get

Well, Once upon a time, I had problems receiving callerid, and then one day, Mark was logged into my asterisk box helping with something else, and I asked him about this, and he showed me a nice tweak to some source file that made it work. Some time later, I must have done hundreds of CVS updates, and along the way, lost the above patch (one liner) and so callerID hasn't worked for a long

For those interested, I'm giving a talk about VoIP/enum.164/asterisk tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS build #2, 4th floor, room 10. Sorry for the late notice, it didn't occur to me that there might be people on this list interested and able to attend etc... -- Best regards, Duane http://www.cacert.org - Free Security Certificates

Guys.. Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? __________________________________________________________________ Anton Krall

We currently have an office which has a happily working Asterisk setup and all is good. We are looking to deploy the same solution to our other 5 offices, my question relates to the interconnections between all those sites. The requirement is seamless dialing of extensions and the ability to have members of multiple offices participate in call queues. I wish to keep the configuration as

works, but there are some issues. I've had asterisk up and running on a suse 9.0 beta 7 x86_64 for about two weeks now. I used then-current cvs, and the compile went smoothly with only a dozen complaints about 64 bit casting of pointer types. The good news : Latency is effectively non-existent (especially when compared to the lame c3 itx box I normally use), and asterisk has not crashed

Dear Duane: Thanks for the steer (presently I route calls to 1800 via iaxtel, but I'll turn that off for that test) I came up with a thought for an interesting e164 service last night - distributed custom ringtones, and custom announcements on a callerid lookup via enum. You'd be able to embed a RINGTONE - an url pointing to either a ringtone definition or a wav file that contains

e164.org is a public name service which provides ENUM.164, a method devised by the IETF and ITU to allow an ordinary telephone to be connected to an Internet type network and provided dialling service from other, regular telephones. Unlike many other "free" voice over IP systems, e164.org allows users who have a regular telephone line, to also hook themselves up to the Internet

Hello all, I am planning to connect my Asterisk with the FWD and/or iaxtel networks. Two mounths ago, I just used the iaxtel network, and i remember I have trouble with this network, I can not place a call. The service do not wotk. With FWD I alwais can place a call, I never get an error from this network. But my experience in both are very short, just a few test time. Could somebody

I'm setting up a private enum zone to simplify/centralize dialplans for a number of Asterisk servers. In several dialed number situations, there are a handful of possible destinations for the call, and I'd like to have * try ENUMENTRY1, ENUMENTRY2, .., ENUMENTRYN just in case the first result is temporarily unable to handle the call. In at least some cases, I'd also like the order in

Anyone using the rate_engine from TrollPhone? There is absolutly no documentation on how to setup data in the tables. If someone could send sample data, or post it to the wiki, it would be helpful. If any others are successfully using another Least Cost Routing method, please pass it along. THanks, Matthew

Can someone send me a quick snippet of a dialplan for international dialing via NuFone? I'm having a hard time getting any help from them this week. Scott

Hello, all. I have some problem when testing my IP phone with Windows Messenger. My IP phone supports such codecs as g729, g711. And Windows Messenger supports red, g711, g723 as you know. The problem comes up when testing with this sip.conf file. ([general] context displayed only) =================================================================================== [general] port=5060

Hi All, A friend has asked me to get a new Zultys 4x5 working with his * server. I've been over the web gui but don't see where all the registration info gets entered in the phone. Is there someone on-list who has Zultys phones and can advise me as to their setup? The manufacturers docs are useless. Thanks, Michael -- Michael Graves mgraves@pixelpower.com Sr.

hi, I am looking to upgrade the firmware on my GS phone but the site doesn't have the IP adress of the TFTP server anymore or anywhere to download the firmware.. Does anyone know this information? What is the current stable firmware version? Later..

Hi, I'd like to setup little private enum server. Any more info on how to do that ? Regards, Rob.

http://advancedippipeline.com/60400413 "BOULDER, Colo. -- Leading Voice over IP service provider Vonage Holdings has complained to the Federal Communications Commission that competitors are blocking the use of its service, according to FCC chairman Michael Powell and others close to the company. "We're very actively on this case and we are taking it pretty seriously," said

Hi folks, I think my little agi script is ready for the big one-oh-oh. Available at http://muware.com/asterisk is cid_rewrite-1.0.0. This agi-script does the following: - Standardize incoming caller-id numbers to adhere to US dialing code; NANPA numbers are reformatted to 1+10, international numbers become 011<country-code><number> (this is customizable with a little PHP knowledge).

My DNS gui(Cpanel/WHM) only allows the following options for entry type: A6 AAAA CNAME MX NS PTR TXT WRK Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? ----- Andrew Thompson http://aktzero.com/

I've looked through the archives - and see questions similar to mine, but no answers. What, if anything, can be done to get the incoming Caller ID to be presented on the Budgetone's Caller ID display? In all other respects the phone+Asterisk seem to be extremely happy with each other.