I have a SIP phone that doesn't support three-way
calling. Is there a way to do three-way calling from a
SIP phone server side instead?
TKS
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I copied all my configuration files for the 7960 to a
new server. I reboot the phone. The phone downloads
the config files from tftp ok. When I lift the
reciever and dial any digit, the phone goes nuts.
Starts flashing and does all sorts of wierd things up
to rebooting.
Any suggestions anyone?
I read the wiki and looked at the examples, but I'm
still having problems. I have a Digium 4 port card
with POTS lines plugged into all four ports. How do I
play the congestion tone the the caller when they try
and dial out but all the lines are in use?
should something like this work?
I have a SIP phone that can make calls but can't recieve calls. Can anyone suggest why?
sip show peers:
Name/username Host Dyn Nat ACL Mask Port Status 601/601 (Unspecified) D N 255.255.255.255 0 UNKNOWN
-- Executing Dial("SIP/206.132.91.139-0814c9f8", "SIP/601|20|r") in new stackJul 18
I have a SIP phone that is registered. i can make calls out from the phone. I can't make calls to the phone.
What does the error message mean? How can I fix it? Thanks!
8 headers, 0 linesDestroying call '[EMAIL PROTECTED]'Jul 15 22:10:49 NOTICE[262159]: rtp.c:285 process_rfc3389: RFC3389 support
There is finally an introductory book about Asterisk!
It looks like Paul Mahler at www.signate.com wrote it
with a lot of help from Digium. I looked at the sample
pages, it looks great.
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We have been using Broadvoice, www.broadvoice.com.
Good price, excellent support.
--- Carlos Arnt [EMAIL PROTECTED] wrote:
-
Hi People,
I wondering here, who is the best VOIP PSTN Provider
to use with my * box ?
That has good prices, good quality (Use ex.
