Hi Gents
I thought I'd pop up and add my 2cents when I just happened across this e-mail
glancing through the list.
I actually do run a production service on Asterisk running on Mac OS X server.
But of course I do it within a VM environment on Linux (Ie Asterisk on Ubuntu
under VirtualBox on
Hi Guys,
Just new to Asterisk and am completely stumped. I have created two accounts
as instructed. Please see below for the config of the user accounts.
[Peter]
type=friend
host=IP address
disallow=all
allow=ulaw
allow=alaw
callerid=Peter 6004
secret=XXX
context=default
Hello,
Try this
[6004]
type=friend
host=dynamic
disallow=all
allow=ulaw
allow=alaw
callerid=6004 Peter
secret=XXX
context=default
port=9060
nat=force_rport,comedia
deny=0.0.0.0
permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0
On Wed, Apr 16, 2014 at
Peter Reid wrote:
Hi Guys,
Kia ora,
Just new to Asterisk and am completely stumped. I have created two
accounts as instructed. Please see below for the config of the user
accounts.
[Peter]
type=friend
host=IP address
disallow=all
allow=ulaw
allow=alaw
callerid=Peter 6004
asterisk-users-boun...@lists.digium.com wrote on 04/16/2014 05:56:32 AM:
From: Peter Reid peter.r...@morodo.co.uk
To: asterisk-users@lists.digium.com,
Date: 04/16/2014 05:56 AM
Subject: [asterisk-users] FW: clients unable to auth
Sent by: asterisk-users-boun...@lists.digium.com
Hi Guys,
Hi All,
Thank you guys - your advice was spot on. I will now reach out earlier and
not struggle with issues like this for 2 weeks J
Best Regards,
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen
Sent: Wednesday,
From the reading and testing I have done it doesn't look like SIP supports
a username and password in the Dial string. I currently have the following
mapping.
priv = dundi-extens,0,SIP,
dundi:pass@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
On the sending side I see
NOTICE[31598]
From the reading and testing I have done it doesn't look like SIP
supports a username and password in the Dial string. I currently
have the following mapping.
priv = dundi-extens,0,SIP,dundi:pass@1.1.1.1/$
{NUMBER},nounsolicited,nocomunsolicit,nopartial
On the sending side I see
Thank you guys – your advice was spot on. I will now reach out
earlier and not struggle with issues like this for 2 weeks J
You sound like you are just getting started with Asterisk. A couple pieces
of advice that helped me when I was starting out:
1. Get a copy of Asterisk: The
I would modify the suggestions slightly and either add to it, or replace
the reference to voip-info with a link to https://wiki.asterisk.org
I was just thinking yesterday that voip-info seems out of date for many
things I search for, but at this point I'm usually looking up things that
are
On Wed, Apr 16, 2014 at 9:06 AM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
I am using DUNDi with SIP to do some least cost routing amongst my various
locations. My mapping is close to what you have:
priv = dundi-extens,0,SIP,trunk_name/number_to_dial
Where trunk_name is replaced
I wanted to move to DUNDi to simplify the setup. It looks like I
need to switch to IAX trunks to be able to do this.
You are a bit outside of what I have done, but this looks like it might be
what you want to do with SIP:
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP--
Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.I configure my Asterisk 11.7.0 to work wit WEBRTC.Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at the Asterisk, but when we try to make a call they send a 488 response and finish it.here is the part of the SIP
On Wed, Apr 16, 2014 at 1:35 PM, Consultor VOIP v...@axys.com.br wrote:
Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.
I configure my Asterisk 11.7.0 to work wit WEBRTC.
Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at
the Asterisk, but when we try to
Hi all,
I have a fresh install of Asterisk 11.8.1 and am putting a Digium TE435 4 T1 card in it for ISDN PRI. I can get the card to be recognised by the DAHDI utilities but when I put in the file "chan_dahdi.conf" with either the generated file from samples with what seem to be appropriate
It's my first post here, so I'll cut to the chase
I have 2 Asterisk servers and want to connect them using sip on one and
pjsip on the other one. One is running at home and another at a VPS. The
first one will be the client (with dynamic ip) and the 2nd the server.
The client uses sip and the
Try starting Asterisk with the -f option. It will NOT fork into the
background so you will see all messages on startup (including any that
might not end up in the log file). Search for DAHDI errors which will
likely be there.
Also, if you configure everything and start DAHDI but don't start
Josh,
Yes, I only have one span currently connected, the other 3 are looped. With the Asterisk process stopped I do see the OK on the "dahdi_tool" screen.
I am not seeing any sort of errors in the /var/log/asterisk directory but when I start asterisk manually with the -f option I do get the
(resend in plain text)...
Josh,
Yes, I only have one span currently connected, the other 3 are looped.
With the Asterisk process stopped I do see the OK on the dahdi_tool
screen.
I am not seeing any sort of errors in the /var/log/asterisk directory
but when I start asterisk manually with the -f
Just a heads up... Enabled NOTICEs on the server and I see this every 10
seconds or so
[Apr 16 18:58:28] NOTICE[2138]: res_pjsip/pjsip_distributor.c:246
log_unidentified_request: Request from 'asterisk
sip:asterisk@179.25.158.95' failed for '179.25.158.95:5060' (callid:
On 04/15/2014 06:52 PM, Kai-Uwe Jensen wrote:
Oops, had it wrong. Here's how it works for me:
[callcentric-template](!)
type=friend
context=from-callcentric
fromdomain=callcentric.com http://callcentric.com
defaultuser=1777xxx
fromuser=1777xxx
secret=password
insecure=port,invite
On 04/16/2014 05:42 PM, Josh Metzger wrote:
Try starting Asterisk with the -f option. It will NOT fork into the
background so you will see all messages on startup (including any that
might not end up in the log file). Search for DAHDI errors which will
likely be there.
Also, if you configure
Greetings,
I want to bond two T1 lines running between a Linux machine with a
quad port tor2 card and a router running Cisco IOS. It seems that the
way to do this is to run multilink PPP across both T1 spans, but the
nethdlc/Linux Generic HDLC stack does not appear to support
multilink PPP.
Is
Enviado desde mi iPad
El abr 16, 2014, a las 6:45 p.m., Sean Darcy seandar...@gmail.com escribió:
On 04/15/2014 06:52 PM, Kai-Uwe Jensen wrote:
Oops, had it wrong. Here's how it works for me:
[callcentric-template](!)
type=friend
context=from-callcentric
fromdomain=callcentric.com
Thanks Johan.
I think I will stick with 1.4.x and DAHDI. Although it is a unsupported
release, I never had any problems with them.
Some machines have never been rebooted for 5+ years.
I am a bit scared of going to 11. I have written a lot of AEL2 script in
Asterisk 1.4.x and I am not sure if
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