similar to: CDR Records are not working

Any recommendations on good wireless SIP phones? Thanks, Pedram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081105/6248fa14/attachment.htm

********************************************************************** Este email assim como os ficheiros que possa ter em anexo s?o confidenciais e para uso exclusivo da pessoa ou organiza??o para o qual foi enviado. Se recebeu este email por engano por favor notifique Redes@bnc.pt Esta nota confirma que esta mensagem foi verificada pelo MIMEsweeper n?o tendo sido encontrados virus.

Dear all i am new to R and using a simple linear model with 4 independent variables and i am wondering if there is a command in R that will give me the partial regression coefficients thanks Pedram Rowhani Ardekani University of Louvain [[alternative HTML version deleted]]

Hi, Having some issues here with getting asterisk realtime for the dialplan (extensions.conf) setup: mysql> desc extensions_table; +----------+--------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +----------+--------------+------+-----+---------+----------------+ | id | int(11) | NO | MUL | NULL |

I saw a couple of posts about this in the archive, but none seemed specifically to address the problem I am having. If I missed something please let me know. Right now I would classify myself as "novice," and there is probably really nothing so trivial that I couldn't possibly have screwed it up. :-) I'm trying to use the AMD command to detect answering machines, and have

Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't need inbound right now) working with FFA and Gafachi. G711 faxing works (as well as can be expected over the internet), but I want the higher reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this: -- Attempting call on SIP/18884732963 at

Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by

We're having an issue where we call ReceiveFax in a context that includes a hangup extension and half the time dialplan execution doesn't continue after the fax is received successfully. Am I missing something simple here? Below is a sample call where this happened: The last log line for this channel/call is: [Feb 21 09:10:53] VERBOSE[13730] res_fax_digium.c: -- Channel

When I call an extension (say my extension 1000)and it goes directly to voicemail the first time, it does exactly what it should do (plays announcement and then records, the second time when i call back (within about a minute), it goes directly to a beep (for recording), no announcement. Another thing, during this time when I call 0 (my voicemail access number) it gives me a fast busy. any

once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you -------------- next part -------------- A non-text attachment was scrubbed... Name: not available

Hi everyone, I've got my * system up and running and I'm really pleased. I've gone with G.711 (alaw) and I've stumbled across a problem; when people place calls internally some people think they have been cut off if the line is quiet for a few seconds. Is there a way of getting comfort noise on the call? I'm using the STABLE release and cisco 7960 phones under FC-1 Cheers

Hi All, I have been reading through the list quite a bit, and I am going to post this more as a poll than anything else. I am working on setting up a very small business with something that resembles a professional voice system. My idea is to use Asterisk with a SIP provider and SIP clients. I currently have a Vonage account already. So adding the 9.99 a month Soft Phone would be easy.

I just signed up with livevoip for 800 DID and have very choppy audio. From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am using IAX and was assigned to server iax01.nyc.*. I do not believe it is a bandwidth problem on my end and I have no problems using iax with gafachi. I opened a ticket with livevoip but no response yet. Would I be better off using sip with them? Is there

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As we are in US and VoiceTrading in Europe, somebody suggested to move the termination minute provider

anyone know a company that will terminate did 831/408 area codes in california. FYI i already checked voicepulse, negative. -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Folks, Randomly, when the phone is taken off-hook, the the Iaxy produces a irritating banshee scream as opposed to a dial-tone. Cycling the power fixes the issue, & sometimes it magically goes away by itself. Has anyone experienced this issue & potentially fixed it? I'm using asterisk CVS head as of jun 17 2004. Thanks, Glen

Hey there, I don't know who else has suffered broadvoices terrible service, but I am about to my end with them. The lack of a LBR codec, the outages, the changing of servers without notifying subscribers haspushed me to my end. Now most incoming calls are abbruptly cut off within a minute of the call starting. Anyone know of any other * friendly providers that have DID, besides Voicepulse,

I am loving Asterisk! I have a Cisco 7960 (Sip) on which I want to try using g729 encoding. I cannot find a setting for this in the phone's interactive screen menu. Do I set it in the sip.conf file? I have also ordered 2 licenses from Digium. My understanding is that because this Cisco phone can handle the encoding, * just passes it thru. Is this correct? Also, I am using LiveVoip for

Hello list, I read on voip-info.org that Asterisk 1.4 support T38 passthrough. So I guess this means that I can have a Grandstream HT503 with T38 support and an analogue faxmachine on the other side of my Asterisk and a T38-account with a ITSP on the other side of my Asterisk machine, right ?! The fax coming from the faxmachine passes the HT503 to my Asterisk and my Asterisk sends the fax to