similar to: [Asterisk-biz] e164.org update

Duane is working on this, there are some problems with asterisk at the moment in delivering this back to the 'caller' and the status ap is the first step in delivering this awareness information. Does anyone want to help duane and matthew write this as they both have like a million things going on at the moment between them. I noticed www.no-ip.com have a dns updater, so something built

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

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:

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

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

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

Stefan Gofferje wrote: > Hi folks, > > I would like to have kinda catchall function for incoming sip > connections. A channel, where everybody could connect to by dialling the > url e.g. sip://guest@<myserver>, like the [guest] section in iax.conf. > I have played around a bit but any attempt to dial any > extension@<myserver> without prior registration leads to a

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

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:

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

Hi guys. I have a XP100P Clone , and the busydetect dont work for me.. PSTN---Asterisk---Sip---Asterisk----PBX Any call from pstn side dont disconnect ... I have no disconnect supervision and busydetect dont work... Please Help me. Zapata.conf [channels] echocancel=yes usecallerid=no hidecallerid=no rxgain=0.0 txgain=0.0 signalling=fxs_ks callprogress=no context=entrada channel=>1

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

Since e164.org added DNC and ADDRESS records my enum configuration has failed. Using both the old EnumLookup app and the new ENUMLOOKUP function, the lookups have consistantly failed since e164.org added E2U+ADDRESS and E2U+DNC records. Mar 6 17:39:44 WARNING[14222]: enum.c:235 parse_naptr: NAPTR Regex match failed. Mar 6 17:39:44 WARNING[14222]: enum.c:354 enum_callback: Failed to parse naptr

Hello, we have set up e164.lu as a test zone, as the delegation for 2.5.3.e164.arpa hasn't been completed yet. For all those who want to call the numbers currently availble directly via SIP, please use the zone name in your enum.conf. If you decide to use the zone, please tell me at mstorck@luxadmin.org, so as soon as the 2.5.3.e164.arpa zone is ready, I will mail you, so you may disable

Hi Asterisk users/e164 developers, I have had an idea on how to improve the functionality of the E164 service. The idea is that when I'm no longer at my house I actually want to receive calls at another number So I have a small software client on my windows desktop and I tell it I'm no longer here I'm going to be on my mobile (similar to the no-ip.com dns updater I use) The client

Does anyone know if the e164.org ENUM service is still active? If anyone who has anything to do with the e164.org ENUM site monitors this list could you check your signup page as the Captcha's (the test to see if you are human) fails for both the text and audio tests every time. I'd post a message on the e164.org forums but the signup page there has the test missing altogether. Greyman.

Hi. I'm using Asterisk & oh323. My gatekeeper needs h323 ID and e164 ID for registration. But I don't know how to send e164 ID. I configure register section in my oh323.conf file like below. [general] <<snip>> gatekeeper=xxx.xxx.xxx.xxx [register] alias=abc alias=777 <<snip>> And the gatekeeper doesn't need passward. It authenticates to use only both h323

ENUM lookups at e164.org return a IP route for ALL toll-free numbers. I was surprised to observe that ALL toll-free numbers get a hit at e164.org. It appears that ALL toll-free prefixes have been delegated, thereby publishing an IP route for YOUR TOLL-FREE NUMBERS, my toll-free numbers, and even toll-free numbers that have not been allocated. :-) See below Should I care? Even though this