9 jan 2012 kl. 09:02 skrev Ronald Cepres:
Hi all,
I've been trying to register a SIP user agent to an Asterisk server using
OpenSIPS as SIP router. The functionality is working fine. However, Asterisk
uses the IP address of the OpenSIPS server as the peer IP address. How can I
use the
Its the sending complete IE. I'm using EuroISDN and I also use overlap
signalling on this interface.
Regards
Hans
On 2012-01-10 01:12, C F wrote:
Exactly which IE message are you trying to push manually? you
shouldn't have to do that, it should be done in the configs for you.
On Mon, Jan 9,
Hi,
1. This patch didn't correct the issue but I'm far from certain that I
correctly applied the patch.
2. I took the Hardware Echocan module off my board and it seems to
correct the issue.
I'll dig deeper to check if I correctly applied the patch and both
report here and in DAHLIN-275 ticket.
On 06/01/12 13:14, Dan Journo wrote:
Is there such a thing as an ISDN30e PCI card which can be used with a
copy of Asterisk, that can act like a voip gateway between the old phone
system, and our asterisk box?
Yes Digium sell 2 port PRI cards that support E1. TE200 series. I use
them like
On 06/01/12 16:17, Ishfaq Malik wrote:
Hi
Does anyone know how to change the target port on a Snom phone.
I have tried adding :new port number to the end of the registrar but
this doesn't work.
It should do. Try putting registrarip:port into Outbound Proxy and
leave the Registrar box just
Can you email me off list (since this isn't really Asterisk related and a
snom support issue, which I can help with) with some details and ideally a
SIP trace?
cheers,
Paul.
Closing this question with a final message including the [SOLVED] phrase
will definitely help the community I
2012/1/10, Olivier oza_4...@yahoo.fr:
Hi,
1. This patch didn't correct the issue but I'm far from certain that I
correctly applied the patch.
I was right to suspect I was wrong : now, after correctly applying
the DAHLIN-275 patch, it's working OK (with the EchoCan module
plugged-in).
Thanks
We have a customer who has asked us to change this behavior, but I haven't
been able to find a way to do it. Server is Asterisk 1.6 and the phones
are SPA 303 and 504.
Receptionist gets an outside call, starts an attended transfer
The office person being called answers by pressing the speaker
You could use a parking lot instead of attended transfer?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Tuesday, January 10, 2012 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Tue, Jan 10, 2012 at 10:24 AM, Danny Nicholas da...@debsinc.com wrote:
You could use a parking lot instead of attended transfer?
Since that actually is more work than just training all the users to
manually hang up, I have a feeling the customer won't be enthusiastic about
it. But thanks
On Tue, Jan 10, 2012 at 12:02 PM, Carlos Alvarez car...@televolve.comwrote:
We have a customer who has asked us to change this behavior, but I haven't
been able to find a way to do it. Server is Asterisk 1.6 and the phones
are SPA 303 and 504.
Receptionist gets an outside call, starts an
On Tue, Jan 10, 2012 at 1:57 PM, Ryan Wagoner rswago...@gmail.com wrote:
On Tue, Jan 10, 2012 at 12:02 PM, Carlos Alvarez car...@televolve.comwrote:
We have a customer who has asked us to change this behavior, but I
haven't been able to find a way to do it. Server is Asterisk 1.6 and the
On Tue, Jan 10, 2012 at 12:01 PM, Ryan Wagoner rswago...@gmail.com wrote:
I did a quick search and found the setting. Go to the Regional tab and
find the Reorder Delay. Change that to 255, which will disable the order
tone and cause the phone to hangup.
Thanks, that did it! Now to
I have been running the windows vovida stun server for some time and has
worked without issue, but I really want to run a linux stun server and get
away from the windows based one. Anyone have an idea of a good replacment
that can be compled on opensuse?
Thanks
Bryant
--
Why don't you just use vovida-linux from sourceforge?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Tuesday, January 10, 2012 3:12 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Linux Stun
Snom is an OEM of the Konftel.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of brya...@zktech.com
Sent: Sunday, January 08, 2012 12:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
So are the Konftel conference room phones any good?
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: Jamie A. Stapleton jstaple...@computer-business.com
Sent: Tuesday, January 10, 2012 5:25 PM
To: Asterisk Users Mailing List -
Hello,
I've been tinkering with Asterisk today for the fun of it, trying to set
up my own domain. I've got pretty much everything working, including a
DID number that connects to my extension.
However, I'm having a problem receiving calls from a particular peer,
specifically my office's PBX,
On Tue, Jan 10, 2012 at 6:00 PM, Christopher David Howie
m...@chrishowie.com wrote:
I've been up and down this issue for a few hours and I cannot for the
life of me determine why simply defining a peer causes Asterisk to offer
telephone-event. I have tried specifying dtmfmode=rfc2833 or
Hello,
I am trying to run load on asterisk server(version 1.8.7.1) for the voicemail()
application using SIPp tool. I am just running sipp at call rate of 1 cps with
the following command:
./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf uac_msg_deposit.xml -i
172.16.129.13 -s
Hi Shalu,
How you are invoking call in dialplan. it's completely depends on that.
And error show that no voice is there for store in voicemail .
On Wed, Jan 11, 2012 at 10:05 AM, shalu dhamija
shalu.dham...@rancoretech.com wrote:
Hello,
I am trying to run load on asterisk server(version
Hi,
Maybe I missed it while checking it, but which spandsp version is
recommended to play with Asterisk 10 and T.38/T.30 gatewaying ?
I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here
(http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a
changelog documenting
2012/1/5, Kevin P. Fleming kpflem...@digium.com:
On 01/04/2012 12:25 AM, Matt Darnell wrote:
Aloha,
We are looking to roll a solution that will have the following network
layout:
ISDN-PRI-- Asterisk-- T.38-- ATA-- Fax
Does version 1.8 with the Digium fax driver have this capability?
Hello,
I am trying to run load on asterisk server(version 1.8.7.1) through SIPp tool
for the voicemail() application. But I am facing a lot of problems. I tried
running 1000 calls from SIPp for 100 subscribers (10 messages for each
subscriber). I am using odbc storage for the messages.
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