similar to: onhold bug?

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

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:

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

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

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

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:

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

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

Is there a way to get the OnHold music to restart without restarting asterisk? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

----- Original Message ----- From: hank To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 27, 2005 9:13 PM Subject: shoutcast mp3 music onhold with amp portal? hello I use amp the asterisk management portal I went to www.voip-info.org I followed the instructions that were given on http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf it isn't working

Anyone can share me with Music Onhold Configuration sample? Thanks in advance for your help, Kang

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

Hello, Anyone using the 1.0.4.55 firmware release with any success? I have had my Budgetone running 1.0.4.50 for about a month and a half now with no problems whatsoever, and I am a little leary about upgrading. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place!