Hi,
I'm trying to connect two asterisk instances using the method described
here..
http://ofps.oreilly.com/titles/9781449332426/asterisk-OutsideConn.html
under the section
Connecting two Asterisk systems together with SIP
I have an user named venu in serverA and vijay in serverB
the serverA
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, 5 May 2013 5:33 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] GotoIf DIALSTATUS - not working
What am I doing
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sandeep Raju
Sent: Sunday, 5 May 2013 8:34 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Connecting Multiple
@Alec,
Thanks.. That was the error.. got it working now.. :)
On Sun, May 5, 2013 at 2:34 PM, Alec Davis siva...@paradise.net.nz wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sandeep Raju
@Alec,
Now I can dial user vijay but the call gets cut after a few seconds and i
get this error in the serverA's console..
http://paste.kde.org/737924
PS: recolgo is the hostname of the system from which I am initialting the
call (using a sip client)
Thanks
On Sun, May 5, 2013 at 2:41 PM,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sandeep Raju
Sent: Sunday, 5 May 2013 9:19 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Connecting
On 04.05.2013, at 20:20, Olivier oza_4...@yahoo.fr wrote:
Le 2 mai 2013 13:23, Marie Fischer ma...@vtl.ee a écrit :
from time to time, we get so-called simplex / one-way audio calls, where one
party cannot hear the other. The only thing in common is that is does happen
with calls via SIP
Le 5 mai 2013 12:19, Marie Fischer ma...@vtl.ee a écrit :
On 04.05.2013, at 20:20, Olivier oza_4...@yahoo.fr wrote:
Le 2 mai 2013 13:23, Marie Fischer ma...@vtl.ee a écrit :
from time to time, we get so-called simplex / one-way audio calls,
where one party cannot hear the other. The only
Mike Diehl wrote:
Is there something I need to do for
the 450 to make this work?
As far as I know, all the Polycoms require a digit map that isn't
blank. You're digit maps are blank. There are two place you can have
digit maps. In the individual phone configs and in the master sip.cfg.
On 05/05/13 20:50, Alec Davis wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, 5 May 2013 5:33 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] GotoIf DIALSTATUS
How to test 911 call?
I'm using Audiocodes and it setup to strip the first number but I've never tested the 911 call.
I don't want to go live as they might charge me.
--
Joseph
--
_
-- Bandwidth and Colocation Provided by
Getting closer...
As a heads-up, the files in the link do not include the locale information
(gd-sip.jar), but I have tracked down something suitable for that...
The phones now get all the files that are essential, but never register
with asterisk (there is no network traffic).
The phone logs
Yes, bad form to follow up on my own post.
Anyway, the secret sauce is indeed a correct SEP.cnf.xml, and a kind
lister provided a working model for SIP 1.9. I now have the endpoint
registered, should be downhill from here. Full write up will indeed follow.
No doubt I will be back soon with
Joseph,
I have made a quite a few test calls to 911. They don't charge you and they
don't get upset.
Just let them know right away it is a non-emergency test call, and then let
them know who you are and what you need to verify on their information screen.
Mark Engelhardt
On May 5, 2013,
If there is a non-emergency number you can call and let them know you would
like to do some test calls. This also allows you to schedule a time for
testing when the PSAP is not as busy allowing for real calls to be handled.
On Sun, May 5, 2013 at 11:15 AM, Mark Engelhardt
I actually work in a 911 center. Please do not dial blindly to do a test
call. Please call the non-emergency dispatch number, ask if it would be ok
to make one or two test calls. If they give you the ok, please complete
those calls as quickly as possible as conditions change in an instant. If
Hi,
I don't know how to call this functionality, but what I want to do is join
an already established communication between PSTN---FXS_connected_phone
using my SIP phone (I have an asterisk v11 with digium TDM400P at home)
Is it possible? What I don't want is using the conference sound and
On 05/05/2013 08:34 PM, neo haux wrote:
I don't know how to call this functionality, but what I want to do is
join an already established communication between
PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11
with digium TDM400P at home)
I had this set up once upon a
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