search for: freenum

The freenum.org beta continues to roll forward. If you have an Asterisk or SER SIP gateway/proxy, please see if you can make some sense of the examples below and install them in your system. Your users will hopefully be able to dial toll free numbers in various nations just like they dial regular numbers in...

I've recently decided to spend idle cycles while waiting for various Astlinux platform builds to complete on making the contents of asterisk/config a little more complete, a little more useful, a little more real-world... I started looking at the possibility of taking JTodd's ISN Freenum cookbook example and updating it: http://freenum.org/cookbook/#ASTERISK_CONFIG but ran into some problems having to do with the following: we use SIP handsets in house, and those operate in the redfish-solutions.com domain... but anonymous SIP (freenum) calls coming in off the internet would als...

Here's a segment of my dialplan, I'm working on the freenum/ISN functionality: same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) same => n,GotoIf($["${isnresult}" != ""]?:fn-CONGESTION,1) ; set up our outgoing call state same => n,Set(SIPFROMUSER=${CALLERID(num)}) same => n,GotoIf($["${GLOBAL(FREENUMD...

The freenum.org project wants to use your trunks! The freenum.org project is an ENUM parallel tree, which has as an eventual goal the distribution of ENUM numbering in nations or areas which due to political or other issues are not able to get secure, inexpensive, or functional ENUM capabilities. As a preli...

I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work "as expected." I would like to be able to do something like: exten => _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with a...

Hi, I've two yocto questions about the syntax of dialplan: 1. What's the "official" notation of each line: "=>" or "=" ? In the wiki of Asterisk, I see very often "=>", however, what's the reason for both syntaxes authorized ? Historical ? 2. To write info in logs/console, you have two commands: NoOp and Verbose. Verbose seems to be

...s are easy to 'visually delineated.' Finding the 'secret' is much easier now. Comparing a 'working' extension with a 'broken' extension will be much easier as well. I use the same formatting in the dialplan. This snippet is from extensions.conf.sample: [outbound-freenum2] exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN}) same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1) same => n,Set(TIMEOUT(absolute)=10800) same => n,Set(isnresult=${ENUMLOOKUP(${EXT...

In several places online, and in the Asterisk F.O.T. book, there is a warning against using '_.' saying: "[it] should probably never be used". However, the need often arises act on numeric extensions that begin with *'s and #'s, and '+', and of course _X. does not match I have tried exten => _[0-9*#+]. but that seems to be the functional equivalent to _X.

...lineated.' Finding the > 'secret' is much easier now. Comparing a 'working' extension with a > 'broken' extension will be much easier as well. > > I use the same formatting in the dialplan. This snippet is from > extensions.conf.sample: > > [outbound-freenum2] > exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN}) > same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) > same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != > "${SUFFIX}"]?fn-CONGESTION,1) > same => n,Set(TIMEOUT(absolute)=10800) > same => n,Se...

...ead process audio frames faster. (Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard) * Add missing 'useragent' field to sip-friends.sql file. (Closes issue #17171. Reported, patched by thehar) * Add example dialplan for dialing ISN numbers (http://www.freenum.org) (Closes issue #17058. Reported, patched by pprindeville) * Fix issue with double "sip:" in header field. (Closes issue #15847. Reported, patched by ebroad) * Add ability to generate ASCII documentation from the TeX files by running 'make asterisk.txt'....

Hello when i dial a toll free no using sipphone i get this error message. How do i solve this? Any help will be appreciated. console message: Starting simple switch on 'Zap/2-1' -- Executing SetCallerID("Zap/2-1", "17473863282") in new stack -- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack -- Executing

I'm helping with an article in "New Scientist" on the use of ISN (http://www.freenum.org/) and the reporter with whom I'm working is trying to get some quotes from users of "normal" ENUM services (e164.arpa, please) from a telco perspective as a comparative basis. If you run an SS7 interconnected telephone switching system that incorporates VoIP via ENUM (hopefull...

FRIDAY September 7th at 12:30 PM EDT http://www.asteriskusersconference.org for more information on how to listen, talk, or both :) This week, ENUM is the main subject, although our friends at e164.org haven't been able to talk to us as planned. Come on by and share what you know about ENUM or ask questions. Also, during Astricon, we are hoping people will call us with reports, either live

.../issues.asterisk.org/view.php?id=15857 https://issues.asterisk.org/view.php?id=12293 https://issues.asterisk.org/view.php?id=11969 https://issues.asterisk.org/view.php?id=11487 and I'd like to do more: * simplify SLA (shared line appearance) through macros, etc. * add E.164 support * add Freenum/ISN support * add diversion support (P-Asserted-Identity, etc) to SIP * add find-me/follow-me via dialplan rules * add telephone auto-configuration hooks etc. if anyone is especially good at troubleshooting dialplan and sip (i.e. extensions.conf and sip.conf) issues, that would be really usefu...

...ead process audio frames faster. (Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard) * Add missing 'useragent' field to sip-friends.sql file. (Closes issue #17171. Reported, patched by thehar) * Add example dialplan for dialing ISN numbers (http://www.freenum.org) (Closes issue #17058. Reported, patched by pprindeville) * Fix issue with double "sip:" in header field. (Closes issue #15847. Reported, patched by ebroad) * Add ability to generate ASCII documentation from the TeX files by running 'make asterisk.txt'....

Anyone wishing to help build/manage openenum.net please contact me via email brian@bkw.org ... I would like to have someone assist in building and management. Thanks, bkw

Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (Context Extension Pri ) State Appl. Data IAX2[iaxtel]/1 ( s 1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 18888476626 1 ) Ring Dial IAX2/XXX:YYYY@iaxtel.com/18888476626@iaxtel But I

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

On Thu, March 26, 2015 22:29, Michelle Dupuis wrote: > You have to consider whether you really want "anonymous" calls, or you > just want to enable SIP calls from trusted companies/partners. The > latter means setting up routes to these companies and (ideally) > registration between peers. > This is what I am trying to get a handle on. It seemed to me that the promise