similar to: mantis password reset link

Hi, I'm Andrew. (Hi Andrew) I'm a toll-free number junkie. I've had an account with iax.cc/sixtel for about a week, and every few days, I find myself sitting at the DID menu clicking the link that reads "Click here to get a random toll free number". I have three toll-free numbers now, and I don't know if it will stop... Is there any hope for me? -- Andrew

Greetings, I'd like to thank everyone that has responded to my original email. I have received information from several companies, and will be testing several of them. I also would like to update a statement from my original message to clarify it: >My strikelist: nufone, voicepulse, iax/sixtel The strikelist is just a list of carriers that didn't meet the needs a resonable

Twice in the last week or so, I've received a message similar to the attached. A portion of the attachment that's attached is not in English. Is this my mail server failing, or someones who's on the list? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ -------------- next part -------------- An embedded message was scrubbed... From:

When you choose to add an unlimited local DID to your account from their control panel, do you get to pick the prefix/NXX, or just the area code? Their isn't any indication of whether or not clicking the Add button will immediately add a number to my account or take me to another screen to pick a NXX. (I don't want to click the 910 area code to find out they giving me a Wilmington or

I am about to start a program that will be generaging sip device configurations for sip.conf. My current sip.conf contains friend entries for each SIP device connected to asterisk. Should I even be attempting to split these in to seperate user/peer devices? Is there(should there be) a convention for referring to the two distinct modes of a device? Can two entries with the same name exist at

--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

> -----Original Message----- > From: Andrew Thompson [mailto:asteriskuser@aktzero.com] > Twice in the last week or so, I've received a message similar to the > attached. > > A portion of the attachment that's attached is not in > English. Is this > my mail server failing, or someones who's on the list? I got one of these, too. It looks like the message I

Greetings, A few weeks ago I posted a message about the upcoming migration from Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been done and all known issues have been resolved. We have scheduled the migration for Sunday, June 5th. The issue tracker will be down most of the day as the migration takes place. Once the migration is complete, the issue tracker will be:

Greetings, A few weeks ago I posted a message about the upcoming migration from Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been done and all known issues have been resolved. We have scheduled the migration for Sunday, June 5th. The issue tracker will be down most of the day as the migration takes place. Once the migration is complete, the issue tracker will be:

My DNS gui(Cpanel/WHM) only allows the following options for entry type: A6 AAAA CNAME MX NS PTR TXT WRK Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? ----- Andrew Thompson http://aktzero.com/

Can someone tell me what this setting is supposed to be? peer->nat = globalnat; It looks like it's inheriting a parameter, but I'm curious, is globalnat an option that we're supposed to set(or let default) in sip.conf? ----- Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close

I saw a few people mentioning they were running or trying to run asterisk on Xen. Last night I checked out v1-0, compiled, ran "make install; make samples", then started asterisk with "asterisk -vvvvc". Several modules refused to load giving this error: [chan_sip.so]May 22 13:48:41 WARNING[4308]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so:

An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m. Para: asterisk-users@lists.digium.com Asunto:

Good <fill in local time of day> I'm looking for a piece of hardware that we can place in two offices that have decent bandwidth, but are in two different US states. There are phone systems on both sides, that have extra CO analog line ports that I'd like to connect through. One side has an IVR, the other side does not use one during office hours. The best configuration would

I've done something similar at home, but made my dialplan such that I can dial either 10 or 11 digits locally. I don't use a "throw away" digit at all. Any 7, 10, or 11 digit call will be appropriately mangled and sent out the PSTN / VoIP provider. ________________________________ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On

I am curious. I understand that features can be pushed to an ADSI phone that make navigating your own voicemail easier, and for other internal things. But, does anyone push this data outside of their own phone network? Example: I am at home with my spiffy new ADSI compatible handset and dial up my bank, OneWorldBankingConglomerate. Would they be able to push me their menu? Button 1 for

Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. -- R.J.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings! Version 1.0.4 of Asterisk, Asterisk-addons, zaptel, and libpri has been released. The releases are available on the Digium ftp site. I have also started a web site called "The Asterisk Development Proxy" where I would like to start centralizing Asterisk development resources. ~ I am hoping that this will become valuable for

I think this is a hole in the documentation that's available right now, so I'm hoping that Olle is watching... help reload says: Reloads configuration files for all modules which support reloading. Well that's all fine and dandy, but which modules actually support reloading? I'm also curious if calls of any known types are affected by the reload. (Does it break anything to do it

There are several ways to know what changes in Asterisk's CVS. This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly up to date CVS changelog summary information. You can also sign up for the Asterisk-CVS mailing list at http://lists.digium.com/mailman/listinfo/asterisk-cvs Archives of the Asterisk-CVS mailing list are at http://lists.digium.com/pipermail/asterisk-cvs/