Hello everybody!
I have some problems with my Astersk. I have an analogical OpenVox card and
A Billion ISDN card (with mISDN).
I load the modules with modprobe zaptel and modprobe wctdm.
When I run ztcfg -vv I have this:
Zaptel Configuration
==
Channel map:
Channel 01:
On Mon, Jun 18, 2007 at 12:50:23PM +0200, Josu Lazkano wrote:
Hello everybody!
I have some problems with my Astersk. I have an analogical OpenVox card and
A Billion ISDN card (with mISDN).
I load the modules with modprobe zaptel and modprobe wctdm.
When I run ztcfg -vv I have this:
Hello, my OpneVox card is an A400P01.
And the output of lspci is:
00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host
Bridge
00:01.0 PCI bridge: VIA Technologies, Inc. VT8235 PCI Bridge
00:0b.0 Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI]
Somebody sugested that we can do this with open VPN .
But somehow. I couldn't do that.
I will be greatful if somebodycan provide more details,
Thanks Regards,
Biju.V.P
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent:
I would like to know if atxfer is supported somehow
because there seems to be little documentation for
this feature. I know most people expect a good SIP/IAX
phone to do the job but I think it's nice to be able
to do attended trasnfers with a simple ATA-connected
analog phone. I have Asterisk
Hello Everyone,
I am trying to install The Asterisk
1.4.5 on Solaris.
Can anyone tell me all the installation steps?
With Thanks,
Gautam___
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asterisk-users
has everyone interfaced Asterisk in a SAP production enviroment?
How?
Thanks to all
--
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
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Have a look at poptop (PPTP server) - pretty straight forward. Great if you
have Windows clients.
If you have Linux clients (or want a permanent tunnel), there are other
options.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit
On Monday 18 June 2007 03:09:40 pm Biju wrote:
Somebody sugested that we can do this with open VPN .
1st Asterisk PBX - install OpenVPN and configure it to run as a server
2nd Asterisk PBX - install OpenVPN and configure it as a client
http://openvpn.net/install.html
The card is a OpenVox
2007/6/18, Josu Lazkano [EMAIL PROTECTED]:
Hello, my OpneVox card is an A400P01.
And the output of lspci is:
00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host
Bridge
00:01.0 PCI bridge: VIA Technologies, Inc. VT8235 PCI Bridge
00:0b.0 Network
Greetings, folks.
I'm having a problem with blind transfers. It seems that, despite not
having the T flag set, callers are able to use the blind transfer option.
Scenario is this:
- Asterisk 1.2.14
- Caller calls into our call center on one of our many phone numbers.
- Call gets placed into
Quoting Dominik Zalewski [EMAIL PROTECTED]:
wouldn't it be simpler just to run voip on some other port that is not
blocked like 80 or 110 etc ?
Then again if your network provider is doing things like that already
what guarantees do you have they are not going to block vpn or
whatever else
We were able to buy a linksys router and install openwrt on it with
openvpn. That router was a client to our openvpn servers at the main
office (where the asterisk box is) and it was able to route all the
traffic that way with a few extra ip table routes. A possible cheaper
solution (since a
Unfortunately, if you don't make the laws (and so few of us do), doing
business in ANY country can be a mixture of good business planning and
good contingency planning. Legislation is always changing, and if you
don't keep an eye on it and have a backup plan, you could very well end
up
On 6/18/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
On Monday 18 June 2007 03:09:40 pm Biju wrote:
Somebody sugested that we can do this with open VPN .
1st Asterisk PBX - install OpenVPN and configure it to run as a server
2nd Asterisk PBX - install OpenVPN and configure it as a client
On 6/18/07, Jon Pounder [EMAIL PROTECTED] wrote:
Quoting Dominik Zalewski [EMAIL PROTECTED]:
wouldn't it be simpler just to run voip on some other port that is not
blocked like 80 or 110 etc ?
Then again if your network provider is doing things like that already
what guarantees do you have
On Mon, Jun 18, 2007 at 02:02:53PM +0200, Josu Lazkano wrote:
Hello, my OpneVox card is an A400P01.
And the output of lspci is:
00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host
Bridge
00:01.0 PCI bridge: VIA Technologies, Inc. VT8235 PCI Bridge
00:0b.0 Network
how can i see that???
thanks
2007/6/18, Tzafrir Cohen [EMAIL PROTECTED]:
On Mon, Jun 18, 2007 at 02:02:53PM +0200, Josu Lazkano wrote:
Hello, my OpneVox card is an A400P01.
And the output of lspci is:
00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP]
Host
Bridge
On 6/18/07, Jay Moore [EMAIL PROTECTED] wrote:
Greetings, folks.
I'm having a problem with blind transfers. It seems that, despite not
having the T flag set, callers are able to use the blind transfer option.
Scenario is this:
- Asterisk 1.2.14
- Caller calls into our call center on one
Thanks for the suggestion.
That is, in effect, what I have done. The difference is that I am using
* as the prefix and not a regular decimal digit (thanks to Leonardo
Kamache).
We have three digit extensions for users and four digit extensions for
system use and special functions. My dialplan
Hi everybody,
I'd like to have the feedback from the community regarding this patch
: http://bugs.digium.com/view.php?id=9972
res_jabber currently relies on the iksemel API to handle TLS
connections, which assumes GnuTLS to be installed on the system. The
basic idea of the proposed
I dont know how to solve your transfer problem, but i have an idea which you
can use to overcome this abnormality.
You should restrict the callers with context which includes only your local
office extensions.
I assume all your incoming calls fall in [default] context. what you should
do is:
I would like to know if atxfer is supported somehow
because there seems to be little documentation for
this feature.
...
I have Asterisk 1.2/Freepbx and
features.conf has a line regarding atxfer and I set it
to *2 (Default). While # works fine (blind transfers),
*2 doesn't.
This was a
nik600 wrote:
has everyone interfaced Asterisk in a SAP production enviroment?
we have integrated SAP systems using JCo and Asterisk-Java. A FastAGI
application accesses to data in R/3 and provides it to the caller.
=Stefan
--
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Hi all,
I have a client that is having problems with phantom calls. I have not
been able to see it happen myself, but they say when it happens, the
display on the phone (polycom 301's) says Device is calling, but when
they answer the phone, there is only silence and then they hang back up
On 6/18/07, Stefan Reuter [EMAIL PROTECTED] wrote:
nik600 wrote:
has everyone interfaced Asterisk in a SAP production enviroment?
we have integrated SAP systems using JCo and Asterisk-Java. A FastAGI
application accesses to data in R/3 and provides it to the caller.
great!
how are the
I experienced the same problem. The only way I could get both
ISDN and analog working was unloading kernel modules for zaptel
and mISDN after boot and then load them in the order:
zaptel first and then mISDN. Still need to find out how to configure
load order in linux.
grz,
Hans Feringa
On
Hi,
I'm using Monitor to record every call is made but I have the problem that
channels are out of sync, for example when some channel ask for something the
answer is heard before the question has ended.
The relevant line with Monitor in the dialplan is:
[EMAIL PROTECTED] ~]# asterisk -r -x
[EMAIL PROTECTED] wrote:
I experienced the same problem. The only way I could get both
ISDN and analog working was unloading kernel modules for zaptel
and mISDN after boot and then load them in the order:
zaptel first and then mISDN. Still need to find out how to configure
load order in
Lee Jenkins wrote:
I have a client that is having problems with phantom calls. I have not
been able to see it happen myself, but they say when it happens, the
display on the phone (polycom 301's) says Device is calling, but when
they answer the phone, there is only silence and then they
At 02:08 6/17/2007, Rilawich Ango wrote:
HI all,
Recently, I got the following message from CLI and finally the
asterisk will hang. Anyone can tell me how to fix the problem or why
it will happen.
Thanks.
Version?
Also:
Hi folks,
I'm experimenting with Heartbeat and whilst I have it running in an
active/standby configuration I cannot get Asterisk to perform properly.
I'm able to start the asterisk software (I imported the aterisk start
file from /etc/init.d into /etc/ha.d/resource.d) with the heartbeat
software
Can anyone recommend any wholesale SIP termination providers that will
automatically charge a credit card? Most seem to want upfront payment
and a credit balance but that sucks when you have to watch it like a
hawk to make sure the balance never hits zero. I'm looking for a
provider that can
Stephen Bosch wrote:
Lee Jenkins wrote:
I have a client that is having problems with phantom calls. I have not
been able to see it happen myself, but they say when it happens, the
display on the phone (polycom 301's) says Device is calling, but when
they answer the phone, there is only
On 6/18/07, Mark Phillips [EMAIL PROTECTED] wrote:
Hi folks,
I'm experimenting with Heartbeat and whilst I have it running in an
active/standby configuration I cannot get Asterisk to perform properly.
I'm able to start the asterisk software (I imported the aterisk start
file from
hi,
I am facing some issues while using MixMonitor. My
extensions logic is attached below:
exten = s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b)
in this extensions TIMESTAMP is not working in Asterisk 1.4. can any
help me why TIMESTAMP is not working in Asterisk 1.4.
Douglas Garstang wrote:
Can anyone recommend any wholesale SIP termination providers that will
automatically charge a credit card? Most seem to want upfront payment
and a credit balance but that sucks when you have to watch it like a
hawk to make sure the balance never hits zero. I’m
On Mon, Jun 18, 2007 at 12:36:10PM -0400, Bob Chiodini wrote:
[EMAIL PROTECTED] wrote:
I experienced the same problem. The only way I could get both
ISDN and analog working was unloading kernel modules for zaptel
and mISDN after boot and then load them in the order:
zaptel first and then
Hi,
Greetings to All,
Im looking for some help on configuring VPN on the Asterisk PBX that I
have hosted in US. Im currently in Middle East and as everyone knows
some countries here has taboo to VOIP. Im not able to get phy phones
registered to my PBX as they are blocking SIP and IAX2.
Asif,
The use of ${TIMESTAMP} in Asterisk 1.4 is deprecated. The new
current method is to use: ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
Therefore your line should look something like this:
exten = s,1,MixMonitor(${CALLERID(number)}-${STRFTIME(${EPOCH},,%Y%m%
d-%H%M%S)}-${UNIQUEID}.gsm,b)
I believe TIMESTAMP no longer works in 1.4. You need to use the below
statement or a variation on it. The documentation does include how to
use this.
{STRFTIME(${EPOCH},,%Y%m%d-%H:%M:%S)}
This will give you the time and date as 20070618-15:36:17.
You can place the variables in any order
The way I have my dialplan set up, the callers shouldn't be able to make
any outgoing calls.
Incoming calls come down my T1:
{zapata.conf}
; T1
group=1
context=incoming_t1
signalling=em_w
channel = 1-24
Which puts them into the 'incoming_t1' context:
{extensions.conf}
[incoming_t1]
#include
On Mon, Jun 18, 2007 at 05:12:57PM +0300, Tzafrir Cohen wrote:
On Mon, Jun 18, 2007 at 02:02:53PM +0200, Josu Lazkano wrote:
Hello, my OpneVox card is an A400P01.
And the output of lspci is:
00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host
Bridge
00:01.0
We were having phantom calls as well. In our case, we had 2 pots line
running in our sangoma card, and when you dial out, would would wait for
whomever to pickup. If you gave up waiting an hung the phone up (we also
had 2 normal phones plugged into fxs ports), it wouldn't immediately
receive the
Dear asterisk users,
I need some help , Im a little new in VoIP , asterisk. I have
downloaded, compiled , installed. I make a simple configuration (Im sorry
write the configuration)
1. sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
allow=all
context=default
register = user:[EMAIL
I too have seen what Rob is saying.. on a Sangoma card. It was an easy fix
in the config, but I don't remember what it was.. but basically it was stray
voltage.
On 6/18/07, Rob Schall [EMAIL PROTECTED] wrote:
We were having phantom calls as well. In our case, we had 2 pots line
running in
Hello,
In your sip.conf you don't have the user for you provider:
[yourprovider]
username=1234
secret=sdfdsf
host=sip.yourprovider.com
type=peer
...
In yor extensions.conf
[mycontext]
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,103,Hangup
exten = 2001,1,Dial(SIP/2001,20)
exten =
try vtund.
http://vtun.sourceforge.net/
its a userland tcp implementation... not the safest thing around, but
should be secure enough for what you are looking for, and pretty
simple to implement.
cheers,
--nvieira
On Jun 18, 2007, at 7:37 PM, Remco Barendse wrote:
Hi,
Greetings to
Vieri wrote:
--- Don Pobanz [EMAIL PROTECTED] wrote:
I would like to know if atxfer is supported
This was a little confusing for me also. A week or
so ago, someone
pointed out that you need to include featuremap in
your extensions.conf
Thanks Don
I'll try that.
It surprises me that
I am new to Asterisk (1.4.5), and I am trying to get chan_mobile working.
My intention is to use it as a cheap GSM gateway.
In the dialplan I configured that all mobile numbers should go thru the
mobile channel. The current situation is that I can setup the call via the
mobile channel
You do NOT want to send realtime audio over a TCP connection.
Nuno Vieira - nfsi telecom wrote:
try vtund.
http://vtun.sourceforge.net/
its a userland tcp implementation... not the safest thing around, but
should be secure enough for what you are looking for, and pretty
simple to
We're dialing a disconnected number via Level 3's vector network, and
are receiving this. The response has SDP in it. Apparently, Level 3 is
playing early media. Asterisk doesn't seem to know what to do with SDP
in a 180 RINGING, and just plays ringing. What am I missing here? How
can Asterisk see
Hello all,
cell -T1- zap -internet-very remote- sip (ip430)
The audio is choppy ONLY to cell USER. The polycom user
says the audio is fine. SIP-SIP calls sound good for both
parties.
Where should I setup the jitterbuffer? The zapata.conf
(recent * 1.2) and/or the polycom configs (fw 2.0.3)?
Matt wrote:
I too have seen what Rob is saying.. on a Sangoma card. It was an easy
fix in the config, but I don't remember what it was.. but basically it
was stray voltage.
On 6/18/07, * Rob Schall* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
We were having phantom calls
The Asterisk development team has announced the releases of
Asterisk-addons 1.2.7 and 1.4.2.
Version 1.2.7 contains some minor updates to the H323 channel driver
that is in this package. Version 1.4.2 contains some additional bug
fixes which include compatibility updates for Asterisk 1.4.5.
Hi all,
Does anybody know why my asterisk doesn't have a show agi command?
Do I have to load any module for it?
Thanks
Ronaldo.
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To UNSUBSCRIBE or update options
I want to setup a blind transer for auto forwarding to SIP peer.
I have context forwarding looks like this in extension.conf
[forwarding]
...
exten = 511,1,Dial(SIP/sip_proxy-out)
...
This will do the re-invite, which is attendance transfer maybe.
But I want a blind transfer by REFER method.
Lucian,
Perhaps this can be of assistance:
http://www.asteriskguru.com/tutorials/transfer.html
-- Alex
On Mon, 18 Jun 2007, Lucian Romi wrote:
I want to setup a blind transer for auto forwarding to SIP peer.
I have context forwarding looks like this in extension.conf
[forwarding]
...
On 6/18/07, Douglas Garstang [EMAIL PROTECTED] wrote:
We're dialing a disconnected number via Level 3's vector network, and are
receiving this. The response has SDP in it. Apparently, Level 3 is playing
early media. Asterisk doesn't seem to know what to do with SDP in a 180
RINGING, and just
On Mon, 18 Jun 2007, Jared Smith wrote:
I could be totally off base here, but it's my understanding that a 180
is telling Asterisk to generate ringing on it's side, and that a 183
(with SDP) would tell Asterisk that the call is progressing and that it
should play the early media specified
Hi all,
Greetings,
Does anybody know why my asterisk doesn't have a show agi command?
Do I have to load any module for it?
It's definitely in there in the source in res_agi.c. My res directories have
their own Makefiles, and it looks like menuselect has an option for
res_agi.c to be
Hi,
I have Asterisk-1.2.18 install with FreePBX more than 75 extnsion,
daily I come accross an issue try resolving them its either user learning
curve or my ignorance.
But, I dont know what to say regarding this issue.
I have my Dial Plan for internal users to have a 3
Hi all.
I've got a project where I need to make outbound calls and play a
prerecorded .wav file to the called number.
So far, I've only been able to make about 15 concurrent calls before the cound
quality gets poor, and I really need to increase this.
I've got QoS configured to prioritize
So far, I've only been able to make about 15 concurrent calls before the cound
quality gets poor, and I really need to increase this.
15 calls isn't very many at all.
I've got QoS configured to prioritize IAX2 traffic above all and my connection
to the Internet is a PtoP 100Mb ethernet link.
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