Displaying 11 results from an estimated 11 matches similar to: "Asterisk WITH Swyx... Any Idea?"
2008 Oct 06
2
Conneting Asterisk to Swyx pri
Hi all, I've done this a few times with other PBX's but swyx has stumped me!
I'm having some trouble getting Asterisk connected to a Swyx system using a
sangoma A104dx... currently the setup is:
BT <-> Swyx
The above setup works fine... what i'm trying to achieve is
BT & SIP Trunks <-> Asterisk <-> Swyx
I have connected to our BT (2 x ISDN30 UK) with
2004 Sep 25
4
Absolutely minimal Asterisk PSTN gateway
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Hello together,
I am setting up a communication server which should also act a
very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500
MB usb memory stick. What is the ABSOLUTE minimum space requirements for
~ running asterisk to work as gateway between isdn and lan? 50MB or 1
GB?(I would compile, configure, etc. on a separate machine
2004 Aug 09
1
Problem moving folders with Thunderbird
I can move around folders to sub-folders in an IMAP account on my
dovecot server, but when I try to move a sub-folder back to the
top-level (by dragging it to the account name), I get an error:
"The current command did not succeed. The mailserver responded:
Mailbox exists."
It doesn't exist on the server though, and I can create a folder with
that name on the top-level.
2004 Jul 28
1
Problems Compiling Asterisk-oh323-0.6.2
Hi.
im compiling the wrapper for oh323(under Suse 9.0)
-pwlib 1.6.6
-openh323 1.13.5. (with oh323 Patch)
i execute:
./samples/simple/obj_linux_x86_r/simph323
and it works fine.
When i Run asterisk-oh323 0.6.2:
make
I get the following errors:
chan_oh323.c:660: error:
`__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__'
undeclared here (not in a function)
2004 Aug 24
1
RC2 and Netmeeting 3.01 ?
Hi,
I'd kindly ask for any guidance how to setup Netmeeting to work with
Asterisk.
I've setup Asterisk as Gateway, selected GSM codec, and I'm able to call
local extensions (no calls into PBX functions) but get no sound.
Any hint, advice ?
Anyone using Netmeeting (maybe also windows messenger) with Asterisk
sucessfully ?
Thanks in advance,
regards,
Robert.
2012 Sep 02
3
Loading Chess Data
All,
What would be the most efficient way to load the data at the following
address into a dataframe?
http://ratings.fide.com/top.phtml?list=men
Thanks,
David
--
View this message in context: http://r.789695.n4.nabble.com/Loading-Chess-Data-tp4642006.html
Sent from the R help mailing list archive at Nabble.com.
2010 Apr 13
1
DAHDI-Linux and DAHDI-Tools 2.3.0 Released
The Asterisk Development Team is pleased to announce the release of
DAHDI-Linux and DAHDI-Tools version 2.3.0.
DAHDI-Linux 2.3.0, DAHDI-Tools 2.3.0, and DAHDI-Linux-Complete are available
for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete
In
2010 Apr 13
1
DAHDI-Linux and DAHDI-Tools 2.3.0 Released
The Asterisk Development Team is pleased to announce the release of
DAHDI-Linux and DAHDI-Tools version 2.3.0.
DAHDI-Linux 2.3.0, DAHDI-Tools 2.3.0, and DAHDI-Linux-Complete are available
for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete
In
2009 Aug 10
0
Vt-d not working with 3.4.1
Hi folks,
currently I try to setup a new xen host v3.4.1 on top of a Asus P5E-VM
DO (latest BIOS, Vt-d capable and enabled in BIOS) to migrate my
extisting HVMs (Win2k3 server) running on Xen v3.3.0 to a new home. I
want to switch over to 3.4.1 to (hopefully!) passthrough my ISDN board
to a HVM domU.
Unfortunate there seem some issue with the VT-d DMAR tables which is
beyond my knowledge and
2004 Aug 13
0
SIP<->H323 "Failed to create smoother"
hello,
Im tryin to make Calls from MS Netmeeting(h323) to
Xlite(SIP) it rings, but as soon as i answered it
dissconnects!!!!
This is what i get from the Asterisk console:
-- Executing Dial("OH323/R27469", "SIP/xlite1|10") in
new stack
Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265
create_addr: Setting NAT on RTP to 0
Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500
sip_call:
2004 Sep 04
1
Oh323, Please Help Newbie ;(
Hi,
I just installed OH323 Plugin and im now tryin to make
simple Configuration to connect Openphone and Xlite to
my Asterisk-Server.
All works fine, i just wanna know if there's a
better way to do it? Is there anything wrong with my
Config?
OH323.conf
[general]
listenAddress=0.0.0.0
listenPort=1720
connectPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=8000
udpEnd=8005
fastStart=no