Displaying 20 results from an estimated 10000 matches similar to: "SCO/Linux concerns"
2004 Aug 23
4
Asterisk WITH Swyx... Any Idea?
Hi,
I'm a student and my thesis work consist in testing
Asterisk with Swyx(SwyxWare).
My approach is to declare asterisk as h323 gateway for
the Swyxserver using oh323 Plugin.
Is there any possibility to connect Asterisk with
Swyx? how?
the outgoing call must pass from Swyxit->to
Swyxserver-> Asterisk->to PTSN
Thanks
2004 May 04
2
Can Asterisk support R2 signaling
Hi All:
I'm a newbee to Asterisk. I currently working on a project and want to know
if Asterisk does support R2 Signaling.
Thanks
Begra8fl
>From: asterisk-users-request@lists.digium.com
>Reply-To: asterisk-users@lists.digium.com
>To: asterisk-users@lists.digium.com
>Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs
>Date: Tue, 04 May 2004 13:32:00 -0500
>
>Send
2003 Nov 24
2
Pressing 0 in Voicemail causes * to hangup
I tried it w/ mine as well and it hung up on me because I just have
Voicemail running not Voicemail2.
It seems as though you have Voicemail2 because it's trying to play the
Unavialable message.
Just a thought though.
Does it do the samething w/
[qout-phillyq]
exten => 0,1,Voicemail(u1)
exten => 0,2,Goto(default,s,1)
Tim Thompson
http://www.amatechtel.com
(806) 722-2227
2010 Jan 10
1
Problem with my dialplan
Hi!
I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk.
I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist.
Any help or any cluees?
Verbosity was 5 and is now 7
-- Starting simple switch on 'Zap/1-1'
==
2004 Jun 17
4
Problems with PRI with T410 messages
Hi all,
I have a box running asterisk with T410 connected to a Nortel DMS 100 switch
and another box running SER with grandstream phones on it
So if there is a call from the pstn it goes from the Nortel to the asterisk
and then to the SER box and finally to the phones.if the phone is busy or
the number is invalid the * box will first send an ALERT message to the
Nortel and say the call is going on
2004 Jun 25
3
Termination Provider
I've been looking for a good iax or sip <==> ptsn provider. Someone
with very low cost usa calling and can offer incoming ptsn connections
in most markets. The only decent providers I could find were
iconnecthere and nufone. Has anyone found someone that really stood
out?
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office:
2004 May 18
0
FW: * and Cisco routers
I understand that softphone are the answer in fact I deploy a ton of the Ip
comm version every week. I am under contract with the phones so I can't
sell them and there no easy way out of the contract.
As for 79XX's I have several office that have them working over a VPN backed
in to our main office where the CCM's and GW's are with managable problem
and for the most part they
2004 Mar 31
2
RE: RxFax/spandsp: not disconnecting
Hi Steve,
I am having this problem in which RxFax is still holding the file after
receiving a complete fax. Somehow the zap channel is still active but on the
fax client it was sent successfully.
If you call the line it is still busy.
Changed from phase 3 to 4
>>> MCF: 8c
HDLC underflow in state 8
Changed from phase 4 to 3
Slow carrier up
<<< DCN: fb
DCN with final frame tag
2003 Sep 17
1
Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
----- Original Message -----
From: <asterisk-users-request@lists.digium.com>
To: <asterisk-users@lists.digium.com>
Sent: Saturday, September 13, 2003 7:55 PM
Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
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2004 Jul 14
1
Digium X100P card to a brazilian analog line
Hello,
I have a problem with connecting a Digium X100P card to a Brazilian analog
line.
Can somebody help me out with this problem?
My /etc/zaptel.conf is
loadzone=br
defaultzone=br
fxsks=1
My /etc/asterisk/indications.conf
[general]
country=br
[br]
description = Brazil
ringcadance = 1000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/4000
congestion =
2003 Nov 12
2
Media Negotiation Failed
Hi, I have this scenario
Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip:
64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than
* )
When a calls comes in Cisco 5300, this send this calls with SIP to *,
asterisk plays a welcome message and resend call to Cisco 3600 that have
4 analog lines connected... but after cisco play welcome message and
when
2012 Mar 10
1
SPA3102 asterisk signaling
Hy all,
Recently a have a little problem with a Cisco device, SPA3102. I use
this device with asterisk to dial out with outbound trunk. (SPA3102
has 1 FXO port)
It working ok , but the device SPA3102 do this : when a call is placed
for outgoing in asterisk and send to SPA3102 , this device "answer
and dial the number in the same time" , in my CLI I see the channel
is open , but on
2004 Jun 10
0
hide caller id
Hi,
We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using
restrictcid=yes and doesn?t work.
What can I do, thaks
Pedro
-----Mensaje original-----
De: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]En nombre de
asterisk-users-request@lists.digium.com
Enviado el: mi?rcoles, 31 de marzo de 2004 12:00
Para: asterisk-users@lists.digium.com
2003 Sep 12
5
Asterisk using a h323 gateway
Hello:
I am testing Asterisk with oh323.
My question is: can Asterisk route some calls thru a second h323 gateway (a
h323 <-> PSTN gw)?
- Asterisk ip: 192.168.1.10
- h323<->PSTN gw: 192.168.1.20
I've tried:
exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20)
or
exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20)
but it does not work at all.
If my h323 client
2003 Nov 27
6
Help for oh323
Hi Friends,
Hope you would help me out here, I have searched the asterisk
user list for hours and also read the readme and test files that
comes with the driver. I need a very simple scenario. I have SIP
clients and want to use oh323 to dial out to PSTN using a h323 gateway.
a)If I set the extention.conf like this:
exten => _87.,1,Dial(OH323/16.52.153.206)
oh323 dials out (I can ring a
2004 Jan 18
1
RE: current version
I tried to use it to create a 'trunk' to Cisco's call manager. The
0.7.1 code worked up to a point.
The call would be established, but audio was one-way from the Call
Manager. Asterisk with
Chan_h323 would not setup the sending rtp stream. The debug results
showed the sending
stream as using ip:0.0.0.0
I have not checked for a CVS update to see if it is fixed, or if that
2004 May 04
1
Probs with oh323 driver: audio only in 1 direction
Hi,
try to setup asterisk as an ISDN2H323-Gateway. The only problem
i have after establishing a call is, that Audio works only from IP to
ISDN-Phone but not from ISDN to IP-Phone.
A known problem ???
Thanks in advance
Michael
i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus)
and oh323-0.6.0
Here are my config's
##############
# modem.conf #
##############
2006 Feb 14
3
consult about Digium Card
Hi All,
I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4
PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?,
other detail is: this card have 4 card green.
I need to know what is the best card for the following scenario: I need a
IVR for my comapny and a PBX, but i want that my extension not use FXS I
want IP phone .
Thanks ins advanced,
2004 Jan 19
2
RE: current version
To be clear I meant using Chan)_h323 with Call Manager where CM is
configured
with * as a H.323 gateway, not client.
CM supports H.323 to direct calls through gateways, and in fact until
recently
that is all they supported. They now also have MGCP, but only to their
IOS
platforms, and SIP is coming soon. There are NO sccp-based gateways,
from Cisco
anyways.
Dan
-----Original Message-----
2009 Aug 28
1
Help needed with getting a maxed-out Asterisk to gracefully deny calls.
Hello Asterisk List,
My company is running a bunch of Asterisk servers behind a Kamailio
(openSER) SIP proxy gateway. Calls come in from our PTSN to VOIP
service to Kamailio, which then randomly chooses an Asterisk server to
handle the call. All Asterisk servers are 1.6.0.9, but the issue I'm
about to describe exists in 1.6.1.5-rc1 as well.
Ultimately what I want to do is cap each