similar to: SIP "catchall"

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

There is now a peering arrangement between e164.org and FreeWorldDialup which means any and all subscribers on FWD are now easily able to make enum calls by prefixing their call with **164, like wise it's almost as simple to make a call to FWD by hitting 8829990<fwd number> This means that for those of you wanting to send/receive calls to/from FWD subscribers you can now do so, easily

For a long time now we've allowed people to publish a wide variety of URI against their enum records such as SIP/IAX2/H323 for VoIP and other types for non-VoIP such as HTTP/MAILTO etc. For the most part these record types aren't listed or aren't utilised so I've done up a quick hack for firefox users as a proof of concept and I'm hoping others will take advantage of this and

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

Firstly we've setup a SIP proxy that uses e164.org to do enum lookups, also rather then issuing people with yet more numbers they have to remember we've coded up a watered down version of e164.org for people that would just like to have a single SIP phone rather then run their own PABX. http://www.Like2Fone.Com for more details on that service. The plan is to get people hooked on VoIP

Long time coming, but we finally have a 3rd party interface on the website to add block of enum numbers in regex form... eg +4412345[678] which will match +44123456 +44123457 +44123458 also +4412345[16-18] which will match +441234516 +441234517 +441234518 or just short prefixes +4412345 so anything starting with +4412345 will match... Currently this is accessible via web interface

And the whole idea of using an enum service is to save those costs and also to encourage intelligent use, If I'm going to call Duane to ask him a question I'm going to call him on his mobile if he's not at home hang the expense but the point is that is wasted money that us intelligent people do not need to spend in the first place. Cheers, Dean -----Original Message----- From:

Soren Rathje wrote: > Next thing will probably be a sbl.e164.org service to block spammers like we do with email... :-) Actually we don't need to do that, using normal NAPTR record can be used instead. We know the IP the call is coming from, we can find out from the NAPTR where calls normally go to based on the phone number, if the 2 don't match filter it. -- Best regards, Duane

You maybe have to create a SIP user called like it is declared in your UAC/UAS xml file. I think it should be 'sipp' or something like that... -----asterisk-users-admin@lists.digium.com a ?crit : ----- Pour: <asterisk-users@lists.digium.com> De: "C. Johnson" <javadude@cedrick.net> Envoy? par: asterisk-users-admin@lists.digium.com Date: 31-05-2004 08:03 Objet: RE:

I'm running the latest version of cvs (not stable), I'm not sure what the other end is running and if this has been fixed or not yet, however I was playing round with onhold earlier, the call went to onhold, and came back from it, then 2 seconds later was hung up unexpectedly, below is what was on console... -- Started music on hold, class 'default', on

Steven P. Donegan wrote: > I don't want to be negative here, but I do believe people who go to do this know the potential risks they face. In many countries (4 of which I have direct, although several year old experience with - all in Asia) taking a local phone line and attaching asterisk to it and gatewaying traffic from other countries will be considered to be 'theft' by the

Grandstream v1.0.4.68 firmware http://www.hellofone.com/downloads.html Seems to have loaded ok on my BT100.. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to

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

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

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

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

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

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).