similar to: VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk

http://www.pcphoneline.com/skype "The VPT1000 is NOT a simple last generation USB phone audio device but is rather a next generation integrated gateway and USB phoneset with simultaneous dual mode Skype and SIP calling support. Skype is not forecast to have "SkypeIn" available until June 2005 but you can have the capability now via its built in SIP capabilities." Is this a

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:

Does Asterisk support Radius accounting?.... -----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 12:08 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs Send Asterisk-Users mailing list

Hi all, I would like people to email me at 'leif at hacklocalhost dot com' some example configuration files for VoIP providers which * can register with. I am going to expand upon the FWD php "wizard" I created for these other providers, but I need some examples as I don't actually use anything but IAXtel and FWD. So far sipphone and iaxtel has been mentioned. I can

Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed

Hi, back in the old analog telephony days there was "digital" PBX-es and digital "system" phonesets. This phonesets have had many individual illuminatable buttons connected with extensions. The PBX can show on the buttons if some extension is ringing (blinks) or busy (constant light), and the user can transfer the call with one touch (pressing one of this button). I search

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

> I'm pretty sure that is incorrect. The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.com

Hi All, I have just compiled the newest version of mpg123 on a RedHat 9.0 system (mpg321 has not been installed) and I am using the newest CVS version of asterisk. Whenever I place any mp3 files in the /var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery death. If mp3s exist in that directory, then I can't even start Asterisk. If I start it without files then copy

I'm just curious if I was to place my * box behind a a FW/NAT box running linux, if my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen -

I live in New Brunswick Canada. The phone company is Aliant. When you set up business service here, you can go with either analog or digital lines. This isn't a T1 or ISDN. They are talking individual lines direct to handsets that they provide. They offer the digital option with even very small ( 2 - 4) number of lines. What technology could this be? Is there any way to connect such a

I'm sorry this is somewhat offtopic, but I do plan to use this to help me create documentation for the * project.. so I guess it is somewhat on topic :) Anyways, I am looking for some sort of document control system. It should act somewhat like a CVS where it keeps previous versions, allows people to submit documentation, keeps track of who has what document open etc.. etc.. The

Grandstream and Global IP Sound have inked a deal in which Global IP Sound will provide its royalty free iLBC codec to Grandstream. GS will integrate this codec into the BT and HT product lines

Hello all I would like change the all announcemennt(Voicemailmain,Voicemail etc.). But I don't know how to change the these each prompt. Do we have any guide book for this? Please teach me about changing the voicemail or other prompt. Thanks

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

In attempting to make my asterisk server the latest and the greatest tonight I attempted to upgrade my CVS. In the asterisk directory I ran the make clean, executed the CVS update -d, and after all the files completed ran make upgrade. My problem is that when I pull up the CLI the cvs version that is showing is the same date as my initial install. Does this mean that the upgrade did not go

http://www.loligo.com/asterisk/misc/Presentations/Asterisk-features-20040106.xls I had been asked a while ago to put together a short Excel spreadsheet listing many of the "common" features of Asterisk as compared to a typical PBX. Many PBX vendors supply an exhaustive list of their features, and I figured I'd take as many of the unique features as others had offered, and put

hi i need many cheaps atas or some very cheap way to connect analogs phones to asterisk what do you recomend? i searches and only find solutions like 40 U$D (in the states, here in argentina is like 80 U$D) per phone any links or something? thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination.

Apart from the "virbiage" FT-201 which hasn't seen the light of day yet, just wondering if there are any other IAX phonesets around available NOW. Don't get me wrong, the Virbiage unit seem to be in the right price range, its just not available yet... Gary .

Hi, I have a strange problem trying to update Asterik on one of my boxes. I have done the following: - delete all the old source files - download the new file using: cvs checkout zaptel zapata libpri asterisk - compiling the new source file using: make clean ; make install Everything is compiled fine, except.... when I stop and start Asterisk server, it is still the old CVS version: