Hi Michael,
You might want to check the voltage settings on the FXS side of
things. Also, are you using the correct signalling? (ground start,
loop start, etc.)
In the Zplex users guide, on page 41 you will see 2 sections on TTLP
and RTLP. That might be of some help to you.
Hey... You have caller
I have had great experience so far with the Snom 190 and asterisk. I
have used them at customer sites and we have them on our desks here at
the office. I have tested the bugettone phone extensivly and I have
had lots of problems with them.
-James
On Tue, 4 Jan 2005 17:15:54 -, Joao Pereira <
Some friendly FYI,I have to say that those are WAY overpriced. My
company also imports those and I know off hand that the single port
version costs $80. They are great boxes, made by welltech, in Taiwaan
(spelling?) and are great ATA devices that work with Asterisk. We put
them at customer sites so
Hi Hadi,
I have been having troubles as well with the FXO/FXS cards from many
installations at customers I have performed. My company has decided to
forgo the FXO/FXS cards and now we use a T1 card with a channel bank.
The versitility and expandibility is tremendous. Plus I enjoy the fact
that I c
Hi All,
Channels 25-28 on a customers PBX are regular Zaptel FXO cards that
are hooked into 4 incomming phone lines. They are all in a group to do
automatic rollover for outgoing calls (if channel 25 is being used,
dial on channel 26, etc.).
Sometimes when a user is dialing a number, instead of br
Hi All,
Channels 25-28 on a customers PBX are regular Zaptel FXO cards that
are hooked into 4 incomming phone lines. They are all in a group to do
automatic rollover for outgoing calls (if channel 25 is being used,
dial on channel 26, etc.).
Sometimes when a user is dialing a number, instead of br
I am suprised that one would have to create a dialplan since its an already built in
function that works with regular POTS phones. Or is it because of the way DTMF is sent
via SIP?
> Someone correct me if I'm wrong but I believe you'll need the dialplan for
> this one...
>
> What I envision is
o
mailbox=9500
callgroup=1
pickupgroup=1
cancallforward=yes
Craig
----- Original Message -
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, August 21, 2004 12:09 AM
Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi All,
I
Hi All,
I am using a Grandstream BT100 and I have been trying to get the PBX features to work
for DND, call foward, etc. These functions do work when I use my POTS phones hooked up
to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP
phones. Is there a feature
returned with error on channel 'Zap/8-1'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Thompson
Sent: Thursday, August 19, 2004 5:23 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP
phone extensio
] Behalf Of Andrew
Thompson
Sent: Thursday, August 19, 2004 5:23 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP
phone extension?
James Freire wrote:
> Sorry about that. I am in the US and using the Digium FXO TDM400 and
> I have enabled a
cupies
space..
-Chris
- Original Message -
From:
James
Freire
To: [EMAIL PROTECTED]
Sent: Thursday, August 19, 2004 12:53
PM
Subject: [Asterisk-Users] Does
Granstream BT100 Conference Button Work?
Hi All, I have
Title: Does Granstream BT100 Conference Button Work?
Hi All,
I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as we
]
Subject: Re: [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP
phone extension?
On Thu, Aug 19, 2004 at 12:07:09PM -0400, James Freire said:
> I have a server setup with an incomming PSTN line and a bunch of
> Grandstream BT100 phones. Is there a way for asterisk to foward an
>
Title: Can PSTN CallerID be fowarded to a SIP phone extension?
Hi All,
I have a server setup with an incomming PSTN line and a bunch of
Grandstream BT100 phones. Is there a way for asterisk to foward an
incomming callerID from the PSTN to the SIP phone that is setup as an extension? We h
sip.conf...can you have 2 @
signs for register?
James Freire wrote:
> Hi All,
> I am trying to setup another sip trunk in addition to what I am already
> using. The sip provider we are using right now gives you your username
> as your email address. So IE. If my email is [EMAIL PRO
Title: Formatting in sip.conf...can you have 2 @ signs for register?
Hi All,
I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is [EMAIL PROTECTED] that is
Title: Can Incomming CallerID be fowarded to a SIP phone extension?
Hi All,
I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID to the SIP phone that is setup as an extension? We have a Voice m
seems that everyone needs different settings dependant upon
many factors...
http://www.voip-info.org/wiki-Asterisk+echo+cancellation
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Freire
Sent: Friday, August 06, 2004 4:12 AM
To: [EMAIL PRO
Hi all.
I am having this echo problem on my SIP phones when I am making a call from SIP to a
PSTN line through asterisk. The echo goes away eventually after a few seconds when the
call starts but it is very aparent during the start of the call. I do have echo
cancellation turned on in asterisk a
/mkdep $(CFLAGS) `ls *.c`
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Oleg A.
Arkhangelsky
Sent: Wednesday, July 28, 2004 9:48 AM
To: [EMAIL PROTECTED]; James Freire
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Trouble compiling asterisk-addons
dif
cdr_addon_mysql.so: cdr_addon_mysql.o
$(CC) -shared -Xlinker -x -o $@ $< -lmysqlclient -lz $(MLFLAGS)
depend: .depend
.depend:
./mkdep $(CFLAGS) `ls *.c`
-Original Message-
From: Oleg A. Arkhangelsky [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 28, 2004 9
Hi Andy,
I have had tremendous success running Asterisk on Slackware linux version 9.1. Its
very quick to install and I had absolutely no problem compiling the source code for
Asterisk or anything else so far. I have asterisk running on 2 servers right now that
use Slackware.
-James
-Ori
Title: Trouble compiling asterisk-addons MySQL
Hi All,
I am having trouble compiling the mysql addon for asterisk. I had downloaded the most recent version from CVS and placed it in /usr/src/ and I get the following error below.
[EMAIL PROTECTED]:/usr/src/asterisk-addons# make install
./m
have included the entire output below.
Thanks a lot!
-James Freire
linux1:/usr/src/zaptel-1.0-RC1# make
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -c -o gendigits.o
gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE
Did you follow the SCRATCH_INSTALL instructions or are you mostly installing
this on an existing system?
MATT---
PS- I wrote the astguiclient suite :)
-Original Message-
From: James Freire [mailto:[EMAIL PROTECTED]
Sent: Friday, July 16, 2004 5:13 PM
To: [EMAIL PROTECTED]
Subj
with php installed also. My guess is that there is a bug
somewhere in the php code but I do not know php well enough to troubleshoot it.
Thanks a lot for any help,
James Freire
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