search for: stormp

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2004 Sep 02
1
why do i get this message emailed to me everytime i post?
Annoying :( S. -----Original Message----- From: Mail Delivery System [mailto:postmaster@bembang.com] Sent: Thursday, September 02, 2004 3:06 AM To: stormp@telus.net Subject: Mail delivery failed: returning message to sender This message was created automatically by mail delivery software (HiveMail). A message that you sent could not be delivered to one of its recipients: sysad The error was: The account has reached its storage limit. -----...
2004 Aug 02
3
How STUN work?
An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040802/d9eed8fc/attachment.htm -------------- next part -------------- ? Hi Can anyone give suggestion why we need STUN while using asterisk behind the NAT. Regards Shan.
2003 Nov 17
2
Static Config?
Hi, I'm new to asterisk. After fiddling a bit I got it to work. It seems great. One question though, is it possible to configure asterisk when it is running? i.e. add new phones or do you have to restart it every time you want to make changes? Thanks, Storm.
2004 Jun 10
1
RE: question about prepaid app_prepaid
Hi, As you asked, I have included my diff to what I did for the DIAL command. I probably didn't stick to some * pre-agreed standard of coding or something, so if these things offend you then I suggest that you close your eyes. :) The biggest thing to consider when you are doing a prepaid system is, what if the person with the same account in/out calls twice? I chose, for now, just to keep
2004 Aug 28
4
incomming call rejected using IAX2 with FWD
Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect attempt from 65.39.205.121 Any ideas? Thanks, S.
2004 Jun 09
1
SIP Registration seems to timeout
Hi, I have an * server on a routable (public) IP address and a sip client behind NAT using a Grandstream phone. He is connected through a bi-directional satellite so he has a bit of latency involved. Usually I can dial this extension and them to me. But I keep getting a registration failed message. I have other sip clients not on a satellite and they don?t get these time outs. So I assumed it