Displaying 20 results from an estimated 400 matches similar to: "asterisk-users Digest, Vol 43, Issue 30"
2007 Feb 25
2
Dialling ZAP channel from analogue
Hi,
Asterisk Version : 1.2.15
Card : TDM11B (1 x FXO , 1 x FXS)
I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP.
The problem comes when I try and make a outbound call.
Here is my extensions.conf :-
Code:
[incoming]
exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1)
exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)
2008 May 22
0
SIP configuration issues
Apologies if this is a repeat: I trawled through the archives and couldn't
find a reasonable answer, so I'm asking here. I have an Asterisk install
connecting from behind a NAT device (DSL modem) to a SIP proxy (in my case,
Broadvoice). I have an sjphone softphone on a Windows PC also behind the NAT
device that connects to the Asterisk install, and using this setup I've been
pretty
2008 Jan 23
7
Asterisk scalability
Hello,
I wonder how Asterisk scales when we increment the Core's or CPU's of
one computer.
I see that Asterisk is only one process (I guess that it uses threads).
But because Asterisk is only one process, this process is always
executed in the same CPU. So we can have a 8 Cores server, with one Core
running Asterisk, another Core running operating system stuff/other
small daemons and 6
2007 Feb 19
2
sip to sip ?
hi all
i've just setup an * box and want to test voip calling, initially from
sip user to sip user...
local sip users can call each other, no issues.
problem arises when i try and call a remote sip account, my * box
always returns "SIP/2.0 404 Not Found"
any ideas ?
2003 Jun 24
0
SIP REGISTER script
Some of you have unusual SIP configurations, and this SIP perl script
may be useful to get remote devices registering with your Asterisk or
other SIP server. Most Cisco routers, as an example, are too stupid
to REGISTER, so this script would be required to dynamically register
them with a remote server. This may not be 100% applicable to
Asterisk, since static registrations are possible,
2010 Jan 29
2
Questions about asterisk and spa2102
Hi there! First mail on the list :)
1.- is it possible to use an spa2102 to make and revice calls from a
"normal" phone? I mean, I know I can use it to connect an analog to an
asterisk server, but I want to know if it can be used to connect
asterisk to the analog phoneline.
2.- I'm trying to unlock the spa2102 with no succes at the moment, any
links or hint will be very
2006 May 02
8
Zapata Telephony interface and torisa module error
Looking at my log file I found the following error:
May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196
May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000
May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded
May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error
May 2 12:00:45 debian
2005 Sep 13
2
passing variables to h extension
Is there a way to pass variables/arguments to the h extension ?
for example :
[default]
exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}},
SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN})
exten => _1098933X.,2,SetVar(_PROVA="bla")
[lot of stuff, agi, goto, tricks and magic that happens]
exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-
2008 Apr 25
2
Cisco to Asterisk migration
Hi Guys,
I have client with a Cisco 2690 call manager solution that wants to upgrade
but cannot stomach the costs of continuing with Cisco
The installation will go up to 100 users
The client currently has about 40 Cisco phones and would like to continue
with these phones with the odd Polycom
I'm looking at plugging in an Asterisk box and using the existing Cisco box
as a PSTN gateway only
2008 Sep 15
4
PBX appliances
Hi List,
Does anyone have experiences to relate on the various Asterisk-based PBX
appliances out there?
Like the Aastra 160, Digium S844i, etc.
Do the Epygi Quadro and Grandstream GXE also use Asterisk?
Thanks,
Femi
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2007 Jul 12
0
No subject
found response from asterisk.
=20
=20
On asterisk's log I see messages like:
"Looking for conference on conference-context (domain serverIP)"
=20
And:
"Call from 'conference' to extension 'conference' rejected because
extension not found."
=20
=20
Does anyone have an ideia of why I'm getting that message?
=20
Why does asterisk seem to be using domain
2010 Jul 05
1
SIP response 482 "Loop Detected"
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same.
Any ideas on how to overcome this problem as we dial
2005 Jan 27
0
Problems making SIP URL outgoing dial
Hi,
I'd like to call my friends through their SIP URLs. I've found two
approaches for doing this in Asterisk:
- one is to prepend some numbers at start and catch them - the rest of
called string is used for SIP URL
- another approach (that I like better) is to use catchall pattern at the
end of context _. and then parse string with help of SIPDOMAIN variable. But
there is a catch into
2006 May 12
0
Sip domains, contexts and CHECKSIPDOMAIN
Hi
I'm struggling with setting up SIP domains.
If I specify a domain and a context in [general], that context overrides
any set in type=peer blocks elsewhere. This results in incoming calls
from PSTN gateways I use arriving in the wrong context.
If I don't specify a context (which the docs I've found suggest is
valid), then I get:
2006-05-12 07:36:16 WARNING[95290]:
2004 Jan 02
3
* Stresstool Help required
Hi all,
I am trying to write a program that sends SIP requests to asterisk. My aim
is to make asterisk record as many voicemails it can at a time. The design
of the program is like this:
There are two processes: One main process and a child process (No flames
pls. I have very little idea about pthreads and dl modules)
The main program asks the user to input the number of test instances. When
2009 Jul 15
2
how to enable dial to alex@asterisk.blurb.com
Hi
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
or even search words for google, as I am not sure how to search for this
type of request.
Alex
--
There is no instance of a country having benefited
from prolonged warfare
-- Sun Tzu - The Art of War
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2008 Apr 03
0
NAT when outbound call leg is not a local subscriber?
Hi,
I have been experimenting with NAT and Asterisk a bit now. Though I have
made progress along the way, I have come across the following problem. I'll
really appreciate if anyone can provide me any help or pointers. Thanks!
Successful Scenario:
-------------------
All sorts of NAT calls are successful with full two-way media when both end
points are locally subscribed users.
Problem
2004 Jan 05
1
Identifying the Originating Cisco SIP Gateway
I have several Cisco SIP gateways sending calls to Asterisk. Because the
gateways don't have user-agents, they don't authenticate with Asterisk. And
because they don't authenticate, they use the default context in the
sip.conf file.
Is there a way to either:
A) identify the inbound gateway with a variable, in channel info, or the
manager interface? If there was a ${SIPDOMAIN} for
2004 Dec 07
1
SIP URLs
I have set up an asterisk server and can successfully call between
extensions using SIP.
i wish to be able to call other sip users using URLs such as
sip:user@sipdomain.com and have no idea how this works... every time i
try it (using X-Lite soft phone), i just get a 404: not found error.
Any clues?
Cheers
Dan
--
Dan Goscomb <dang@cashcade.co.uk>
2005 May 22
0
Digium and IPsando announces agenda for Astricon Europe - register now!
The agenda for Astricon Europe in Madrid June 15-17 is now coming
together. It will be an international conference, with speakers from
both USA and Europe. Last year, we had over 25 nationalities
participating in the first Astricon - the Astricon where Mark released
Asterisk 1.0 on the conference floor, during his keynote!
Many active members of the Asterisk community talks at the conference,
one