similar to: E164 caller ID

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

So I just saw this VoIP-centric article at slashdot (http://slashdot.org/article.pl?sid=04/05/22/1840220) which mentions e164.org. It's a "non-profit public DNS root designed to map phone numbers to Internet protocols." Is anyone on this list actually using this? They have asterisk config instructions: http://www.e164.org/config.php I wonder if someone can help me understand

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

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

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.

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

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

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

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

Hi Friends, i have installed and configure asterisk-1.8.0. When i have tried asterisk start get below errors and not able to start asterisk. *FD 32767 exceeds the maximum size of ast_fdset!* Thanks in advance. -- Best Regards, Rajnikant Vanza -------------- next part -------------- An HTML attachment was scrubbed... URL:

i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part --------------

We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone<=>SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a

If I have the following in the dialplan exten => foo,n,Dial(SIP/1234&Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension

Hi, I feel I've read a thread about this previously but I couldn't find it. Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. So that, you could

Imagine that you have this code: exten => _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})})) If ${QueueName} happens to be unset, this will cause a warning: [Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187 queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an argument: queuename The obvious solution: exten => _X!,n,ExecIf($["${QueueName}" !=

Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing "bs523450017" instead of +6523450017. i tried putting it inside double quotes CALLERID(num)="+6523450017" telco says the same thing. is this possible? thank you Regards, nhadie

I have a problem with asterisk-1.2.13, where it retries SIP INVITEs too quickly. It happens when qualify is on, and the server it tries to reach is only 1ms away according to qualify. The time between the first SIP INVITE and the 7th (last) is then only 64ms, and that can be too short for the peer to react. I reported this bug in much more detail in bugs.digium.com, but the bug is gone now

Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. Thanks

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

I have two Asterisks connected using SIP. One is acting as a SIP "server", the other as a SIP "client". This almost works; but calls from 50607795 are rejected with this error: check_auth: username mismatch, have <50607796>, digest has <50607795> On the "client" I have these accounts configured in sip.conf: register => 50607795:test at