I have been using Asterisk 1.0-RC2 successfully with a channelized T1
circuit for quite a while now but after upgrading to 1.0.0 callerid no
longer works properly.
Debug output from a channel shows what actually is received through DTMF
from the carrier:
<< [ TYPE: DTMF (1) SUBCLASS: * (42) ]
I don't know that Plantronics stuff qualifies as inexpensive but I have
been using Plantronics H headsets with the adapter at this link.
http://store.yahoo.com/founderstelecom/dirconcabfor.html
I have two of these cables and they work very well.
Bryan
Nate Carlson wrote:
Sorry, I know it's OT, bu
I have been adding site:lists.digium.com to my Google searches and that
seems to work well.
Bryan
Muiz Motani wrote:
This brings up a good point that has had me scratching my head for a long
time. Is there a good searchable archive of the asterisk mailing lists? I don't
particularly want to dow
It should be above the channel line. The callerid line is not included
because it is below the channel line.
Bryan
Ronan wrote:
That does not work for me. I had tried that, but no luck. This is what
I have in there for it.
context=darby
usecallerid=yes
musiconhold=default
echotraining=yes
echoc
From what I have been told and my own experience, you cannot set the
7960G's switch port to be in a different VLAN unless you are using a
Cisco switch. The way it is set on a Cisco switch is through the
following configuration parameters for an individual port:
switchport mode access
switchport
Because of all the troubles I am having with the TDM400P cards working
consistently for outgoing calls, I am trying to setup a Cisco router as
my pstn gateway. The router is a 1751-V with a VIC2-4FXO card installed.
I followed the examples at the following two links to get it setup:
http://www.
I had this problem with 7.1 initially because there are two different
filename schemes for 7.1. One is P003-07-1-00 which should be listed in
the file OS79XX.TXT on the tftp server and also P0S3-07-1-00 which
should be in SIPDefault.cnf on the tftp server as follows:
image_version: P0S3-07-1-00
I am trying to run asterisk on Trustix 2.1 w/updates. Has anyone had any
troubles related to zaptel hardware like sporadic errors when trying to
dial out? Also, what is the consensus for the best Linux distribution to
run asterisk on? Trustix seems like a nice choice but I would like to
hear th
I am running asterisk 1.0-RC1 with zaptel 1.0-RC1 on Trustix 2.1 on with
a TDM400P with one FXS and three FXO modules. I am having intermittent
problems when I try to dial out from any of my Cisco 7960 or 7940 SIP
phones. I have my two analog lines configured in zapata.conf as follows:
signalli
