I am having a problem where gateways require authentication which
works perfectly for standard calls, however when a gateway failure
occurs the next gateway fails auth.
I can see that the packet contains the Authentication header which
fails as the security relate to the previous gateway.
How
I've seen the 'uac_auth' option in OpenSIPS too. The question is how
to use it :-)
On 14 September 2010 12:41, Andrew Pogrebennyk
andrew.pogreben...@portaone.com wrote:
On 14.09.2010 14:30, Ross Beer wrote:
I am having a problem where gateways require authentication which
works perfectly
:
consume_credentials()
On Sep 6, 2010, at 6:23 PM, Ross Beer beer.r...@googlemail.com wrote:
Hi All,
I have a question relating to Asterisk call pickup. I would like to
pass a request to all gateways and whichever gateway answers the call
wins.
Should the request be sent to all servers
, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
Hi Ross,
To use get_dialog_info() and dlg_val's you must create the dialog before
(see create_dialog() function in dialog module) - otherwise all dialog
ops are invalid.
Regards,
Bogdan
Ross Beer wrote:
Hi,
I am using the following piece
module) - otherwise all dialog
ops are invalid.
Regards,
Bogdan
Ross Beer wrote:
Hi,
I am using the following piece of code to set dlg_val and then using
get_dialog_info to check to see if a user already has a call and if so
pass the new call to the same gateway for attended transfer
Hi,
I've been trying to use the
get_dialog_info(callee,$var(x),caller,$fu) to make calls go
back to the same gateway as they are already active on. However when
setting a '$dlg_val(caller)' variable after load balancing stops the
load balance and just selects the first gateway.
For example, I am
Hi,
When using load balance $dlg_val(caller) = $fu; the load balance
only selects the first gateway.
I try to check for an existing dialogue and if one exists route to the
server address stored as a dlg_val which works fine, but calling
load_balance after setting the new dialogue values only
Hi,
Is it possible to use the next gateway from within a failure route
when using the load balance module?
Thanks,
Ross
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Hi All,
Is there a way I can send a 'call pickup' request to multiple Asterisk servers
at the same time, then the first one to answer gets the call?
I think this is some sort of fork or branch?
If anyone can offer any advice or examples I would be very greatful.
Kind regards,
Hi,
I currently have a problem with one way audio or no audio at all on some calls
between a Snom and an XLite client. Both are on different networks.
I am using nathelper to resolve natting issues with SIP packets and MediaProxy
for all calls using:
engage_media_proxy();
It
Hi,
I don't know if anyone could point me in the right direction, but I am trying
to get Presence, BLF and BLF dialog working so that when a call is made the
lamp lights show the staus on the Snom.
At present it all works for a few minuites and then either just stops working
or the
-system.ro:
Hi Ross,
It is essential to understand what transfer method you are using (to see
how to catch it and force it to the same gw) - REFER based ? or ?
Regards,
Bogdan
Ross Beer wrote:
Hi,
Is there a way to direct calls from a user based on an active
dialogue, for example record
Hi,
Is there a way to direct calls from a user based on an active
dialogue, for example record what gateway a call originated from and
then if the called user transfers a call send the request via the same
gateway?
I have seen:
set_dlg_profile(caller,$fu);
Can this be used to identify a
Hi,
Is it possible to replicate presence between two servers? I have a MySQL
cluster that replicates all of the tables except location as this is handled
at SIP level.
The problem I can see is that the active_watches table has a 'socket_info'
field that stores where the user subscribed which
Hi All,
Is there a way to share presence across multiple servers. For example if I
implement presence using a mysql cluster will each machine know about the
other's presence status and update clients automaticaly?
If not, what is the best way of sharing presence?
Thanks,
Ross
Hi,
I have installed openxcap however when I try to run it I get the following
error. I have installed twisted, twisted-core, twisted-web and twisted-web2
however the error states it can't find web2.
Can anyone point me in the right direction?
Thanks,
Ross
Ok, I managed to get the module dependencies resolved however when running
OpenXCAP I get the following error:
openxcap --no-fork
Starting OpenXCAP 1.1.2
/usr/local/lib/python2.6/dist-packages/xcap/tweaks.py:1: DeprecationWarning:
the md5 module is deprecated; use hashlib instead
import md5
You need to add a path to a sip message and then replicate at sip level. The
code I use to do this is:
if (is_method(REGISTER)) {
# Uncomment this if you want to use digest
authentication
if (!www_authorize(, subscriber))
() a couple of places in your
config depending on what you want to accomplish.
Ross Beer-2 wrote:
Hi,
I am using MediaProxy to help get over some one way audio issues, however
it
appears to be causing more problems than it is fixing.
When I make a call between two registered phones
Pascu d...@ag-projects.com
On 27 Oct 2009, at 18:24, Ross Beer wrote:
'caller_bytes': 482400, 'callee_bytes': 494400
According to this excerpt from the statistics, the relay got packets
from both sides. You should check your ip_forward settings and the
rp_filter settings (1st should be on 2nd
Hi,
I am using MediaProxy to help get over some one way audio issues, however it
appears to be causing more problems than it is fixing.
When I make a call between two registered phones there is no audio at all,
but when I call a gateway audio passes correctly.
Looking at the logs it indicates
I am having a problem getting phones to talk to mediaproxy. I get oneway audio
however when I dial an asterisk server and do an echo test audio passes
correctly.
I can see that RTP streams are not reaching the mediaproxy even though the SDP
is updated to the correct addresses. Firwall ports
should work
correctly.
This is very confusing!
Ross
Message: 6
Date: Sat, 24 Oct 2009 06:56:27 +0200
From: Sa?l Ibarra
Subject: Re: [OpenSIPS-Users] One Way Audio Using X-Lite, OpenSIPS
Asterisk (Ross Beer)
To: OpenSIPS users mailling list
Message-ID
Hi,
I am using the following config with Opensips and having a problem with one way
audio. When connecting the softphone directly to asterisk that runs on the same
machine audio passes without any problems. Firewalls are all open and
Zoiper/Snom phones connect without issue.
If anyone
0016e640d47e2c8bcd047677c...@google.com
Content-Type: text/plain; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable
MIME-Version: 1.0
Hi Duane=2C
=20
Here is my config and SIP trace.
=20
There is one interesting thing in the SIP trace=2C that is the SDP codecs. =
I have G711u=2C
I have a server located on the internet running opensips and asterisk. When
registering directly to asterisk I can perform echo tests and make calls.
If I register to Opensips and use the load_balance there is one way audio. I
can hear sounds coming from the asterisk server but sound from
Subject: Re: [OpenSIPS-Users] One Way Audio
From: duane.lar...@gmail.com
To: ross_b...@hotmail.com
Are there any firewalls or NATing involved?
On Oct 21, 2009 10:13am, Ross Beer ross_b...@hotmail.com wrote:
I have a server located on the internet running opensips and asterisk
2009 15:23:04 +
Subject: Re: [OpenSIPS-Users] One Way Audio
From: duane.lar...@gmail.com
To: ross_b...@hotmail.com
Are there any firewalls or NATing involved?
On Oct 21, 2009 10:13am, Ross Beer ross_b...@hotmail.com wrote:
I have a server located on the internet running
to?
Whats the destination? Is it actually sending the RTP to the Asterisk box?
On Oct 21, 2009 11:15am, Ross Beer ross_b...@hotmail.com wrote:
Yep, traffic comes from the asterisk server and can be heard on the
softphone, but when the echo test starts no audo can be heard
Hi,
I have registrations replicated over two servers using a MySQL cluster and have
also tried replication at SIP level however I can not call between servers.
The setup is as follows:
Soft phone A Reg OpenSips A
|
Hi,
I have two OpenSips servers setup and each server replicates its registrations
over to the other server.
At present I am not able to call between servers, i.e. a soft phone registers
to server A and another to server B, but A can not call server B even though
the registrations are
the phone registers too.
Thanks,
Ross
What error message are you getting?
On Thu, Oct 15, 2009 at 1:41 PM, Ross Beer ross_b...@hotmail.com wrote:
Hi,
I have two OpenSips servers setup and each server replicates its
registrations over to the other server.
At present I am
To: ross_b...@hotmail.com
Subject: Re: [OpenSIPS-Users] Problem Running MediaProxy Build
Date: Tue, 28 Jul 2009 23:09:31 +0300
CC: r...@ag-projects.com; users@lists.opensips.org
On Tuesday 28 July 2009, Ross Beer wrote:
That has give me:
module 'gnutls' from '/var/lib/python
Hi,
When running the MediaProxy ./setup build I get the following error:
---
./setup.py build
Traceback (most recent call last):
File ./setup.py, line 7, in ?
import mediaproxy
File
, Ross Beer wrote:
Hi,
When running the MediaProxy ./setup build I get the following error:
---
./setup.py build
Traceback (most recent call last):
File ./setup.py, line 7, in ?
import mediaproxy
File /home/voicehost/MediaProxy
the
wrong one (the older). You can verify what it finds by default and
where it is located by running this:
python -c import sys, gnutls; print gnutls.__version__; print
sys.modules['gnutls']
On 28 Jul 2009, at 21:04, Ross Beer wrote:
Hi,
I have compiled and installed mediaproxy
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