similar to: hang up when going to voicemail

GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can

Message: 16 Date: Thu, 19 May 2005 00:16:34 -0600 Michael, Do both! As for Sip Termination: ----------------------- Contact Kristi Eggers @ Txlink.net for month to month Originating/Termination VoIP Toll Free or Local USA DID #s. Yes they do both Sip and IAX. You must have seperate accounts for either Sip or IAX and fund your account with a minimum of $100. This is what I did. Once I get

BJ, >BJ Weschke <bweschke@gmail.com> >Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom >SIP termination vs. DS3 >To: Asterisk Users Mailing List - Non-Commercial >Discussion <asterisk-users@lists.digium.com> >Message-ID: <79cf63305051908056c284cc9@mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 >Did I miss pricing/availability

Mike, No they have not. Calls are failing again today. They have offered to refund my money but that does not solve the problem. My asterisk server is only 4 to 12 ms away from their "network". I have had VERY good luck with nufone.(40 to 45ms away) Only have 1 or 2% fail rate. Going to be calling txlink.net on Monday. Seems that LiveVoIP does not care about asterisk users. They like

I launched www.goiax.com last week, which is intended to promote the use of IAX as a free and open source alternative to products like skype. There is no charge for the service. Right now I have free outbound to united states toll-free and us domestic numbers working. Currently the site hands out a virtual 87820-xxxxxxx number but I intend to add the ability to get a free United States DID

help me test load a box! I have a new box with four PRIs on a TE405P I will terminate US Toll-free traffic (1-800, 888, 877, 866) for free via SIP to anyone who wants to test. Just email me at matthew@txlink.net if you would, to let me know that you're testing, and with any comments about quality, etc. I have ulaw, alaw, and GSM codecs enabled. To use, just send your call via SIP to

Here is some code to do authentication by caller ID for DISA through AGI. My original code had a bug in the Mysql query code, and there was a hangup in the wrong place [that's what I get for coding something at 2:00am], but the attached code works correctly. Take note of the REGEXP for the CallerID variable. When I tested the code from the PSTN it worked because there was no name component,

Dear all, Can I do a permutation test if the number of individuals in one group is much bigger than in the other group? I searched the literature but I didin´t find any assumption that refers to this subject for permutation tests. Best regards João Fadista Ph.d. student UNIVERSITY OF AARHUS Faculty of Agricultural Sciences Dept. of Genetics and Biotechnology Blichers Allé 20, P.O.

Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99 retail. They have a version with a router for $89.99. We picked the non-router version up and it seems to be a rebadged Sipura SPA-2000. The box has a Vonage service package inside as well, but it does work with other services. The box also has a "User Guide" meant for end-users that is very well written [no

HI All, I want to consume the SOAP apis using SSL in the SAVON gem, but I am facing the below error while trying to access the soap services. client = Savon.client(wsdl: "https://xxx?WSDL") client.operations HTTPI GET request to xxxxx (curb) HTTPI::SSLError: Curl::Err::SSLPeerCertificateError In the Savon site <http://savonrb.com/version2.html#globals-ssl> I found that options

Hello list, I have about 20 Digium TE-405Ps out in the field, and I started having trouble with one just recently. The card had worked fine for a month with 4 PRIs in NFAS configuration, and then all of a sudden I started getting a disappearing D channel. A restart of asterisk / ztcfg /module unload-load did not fix the problem, but a reboot [not power off, just a restart] would bring it

--On Thursday, February 03, 2005 2:20 PM -0500 Andrew Thompson <asteriskuser@aktzero.com> wrote: > What you are seeing with these bargain providers is they have a clause in > their contract that says they own the number, not you. It is a lock, and > it ought to be illegal, but sadly, it's probably not. If you choose one > of these companies that doesn't allow you to

anyone have a recommendation for a place I can buy cheap motherboards that supports those 64-bit 3.3 volt PCI slots for the T100P ? I can't find them at Fry's or anywhere locally. All I can find online is dual processor server boards that are overkill for this application. I would like to use a P3/ P4/ AMD single processor. No Xeons or dual processor junk. Anyone know why digium

What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers.

Hi there, I've made a couple of small contributions to the wiki but recently I read the Terms of service, they are pretty draconian: LICENSE AND SITE ACCESS voip-info.org grants you a limited license to access and make personal use of this site. This license does not include any resale or commercial use of this site or its contents. Without express written consent of voip-info.org you may

Hi. I read most of the postings here but unfortunately I didin''t found a complete example which could be used. Of cource the ones who are professionals in javascript could implement the missing peaces from the puzzle. What I''m required to do is a tree (sortable list) where items can also be deleted and at each modification a function (ajax) is called to save the changes. For

Hello list I'm having some trouble getting * to recognize ztdummy on a new install (Fedora Core 3). I installed the module, and it shows up when I do an 'lsmod': Module Size Used by ztdummy 3796 0 zaptel 206852 1 ztdummy crc_ccitt 2113 1 zaptel I did 'service start zaptel' and everything appears cool.

Message: 5 Date: Thu, 19 May 2005 21:44:11 -0500 From: "Matthew Simpson" <matthew@txlink.net> Subject: [Asterisk-Users] cisco 7960 question To: <asterisk-users@lists.digium.com> I have a stupid question. How do you remove line presentations on a cisco 7960 ? I have 3 line presentations I don't use anymore and I can't figure out how to remove them. If you look in

Mike, Martin O'Shield here from WindyCitySDR. >From: mike castleman <mlc@democracynow.org> >Subject: [Asterisk-Users] origination providers >To: Asterisk Users Mailing List ><asterisk-users@lists.digium.com> >Message-ID: ><20050524190016.GG11096@pinetree.mlcastle.net> >Content-Type: text/plain; charset="us-ascii" >hi folks, >Has

I am trying to use chan_skinny but when loading the module I get: [ Booting....../usr/lib/asterisk/modules/chan_skinny.so: undefined symbol: ast_pickup_call I am using CVS 07/23 I can't get chan_sccp2 to compile, it gives me parse errors, or I'd be using that. :-/