info to the
DB I would hope the 2nd server would have access to the system state at the
time the first server became inactive. You would just lose whatever calls
ended during the period between 1st server failing and 2nd starting (max 20-30
seconds in my experience).
Kind regards,
Mark
/RTpproxy
ßà Asterisk
9.9.9.9 8.8.8.8 external/internal 1.1.1.2
1.1.1.3
So I need traversal and translation over same interface in both directions ?
(seems like a b2bua, but only need load-balancer for this scenario)
Hilsen / Regards
Mark Henriksen
Mark Henriksen would like to recall the message, "[SR-Users] Kamailio B2BUA
or Dual router ?".
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ot; and can not read the Wiki well enough J
Hope you can help
//Mark
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, as when I run the same config with 3
IP's, but NOT using VRRP/Keepalived/Aliased IP's, everything works normally.
However, I do need to run this setup in HA, so would welcome any suggestions as
to how I might resolve this issue.
Kind regards,
Mark Hall
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I do not understand your references to sysctl - can you provide any example?
Many Thanks,
Mark Hall
- Original Message -
From: Olle E. Johansson
To: Kamailio (SER) - Users Mailing List
Sent: Friday, February 13, 2015 9:06 AM
Subject: Re: [SR-Users
carrierroute cfg
make group_include=db mysql standard all
make install
I have installed libconfuse-devel and the tm module is loaded.
Would welcome any helpful suggestions.
Thanks,
Mark Hall
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http
experiences so far?
Thanks
Mark
On 19 Jun 2014, at 09:50, Jonathan Hunter hunter...@hotmail.com wrote:
Hi All,
As you guys might remember I have been doing alot of work with these legacy
handsets recently, in particular the Cisco IP phones 7945G and 7965G.
They work well now with kamailio
if I’ve got something wrong in the .cnf xml …
Cheers
Mark
On 19 Jun 2014, at 10:48, Jonathan Hunter hunter...@hotmail.com wrote:
Hi Mark,
Sure of course, they are some what painful to get working due to their
asymmetric NAT behaviour.
What handset models are you working with, and what
IP in the proxy field doesn’t seem to make any
difference.
Cheers
Mark
On 19 Jun 2014, at 11:51, Jonathan Hunter hunter...@hotmail.com wrote:
Hi Mark,
Yes we had exactly the same issue with a 7940.
It is with the .cnf xml file.
You need to make sure that the line name is populated
I have a scenario where calls that are transferred to a shared line doesn't
trigger the in use light to show up on both phones that have the shared line.
but when i make an outgoing call, let's say, using the shared line, the in
use light lights up on both phones that have the shared line.
having to do with INVITES
to see if it puts me on the right track.
thanks
From: Daniel-Constantin Mierla mico...@gmail.com
To: mark li limar...@yahoo.com; Kamailio (SER) - Users Mailing List
sr-users@lists.sip-router.org
Sent: Wednesday, April 2, 2014 5:31:33 PM
Hi there.
I've noticed that no matter what i do with my test phone (call voicemail, call
another extension etc) I get error messages like the following in my log file:
Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace:
c=[/etc/kamailio/kamailio.cfg] l=553 a=26
I'm using kamailio as a sip proxy and registrar, and freeswitch for all media
services like vmail, conference calls, ivrs etc.
I just set up vmail ... and it seems to be working. Except that it appears
that I've broken something else because now when I call extB from extA, instead
of ringing
of
debug data on the console and it's hard to scroll through.
thanks.
From: mark li limar...@yahoo.com
To: Olle E. Johansson o...@edvina.net
Cc: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org
Sent: Friday, March 28, 2014 8:47:51 AM
Subject: Re
Here's a copy of my cfg file: http://pastebin.com/GtDwWKWr
I tried to run with the switches you mentioned... but i still don't get
anything logged to syslog.
?
sorry... and thanks for your patience.
From: Olle E. Johansson o...@edvina.net
To: mark li limar
Hi there.
I'm still trying to integrate Kamailio and freeswitch... where kamailio acts as
a proxy and registrar ... and freeswitch provides conference calls and
voicemail.
I have calls between two polycoms working and conference calls work.
But when I try to leave a voice message for a user
to be for
connecting calls between extensions.
I guess I will have to revisit it to make it more specific... perhaps to ignore
calls that come in with a vb- prefix... or any prefix for that matter.
Thanks. And sorry for the noise.
From: mark li limar...@yahoo.com
To: sr
in the acl.conf.xml file...
just wondering if kamailio has something similar?
Thanks.
From: mark li limar...@yahoo.com
To: mark li limar...@yahoo.com; sr-users@lists.sip-router.org
sr-users@lists.sip-router.org; Kamailio (SER) - Users Mailing List
sr-users
loadmodule
xlog.so
and i've also added a modparam for the log like so:
modparam(xlog, buf_size, 8192)
Thank you.
From: Olle E. Johansson o...@edvina.net
To: mark li limar...@yahoo.com; Kamailio (SER) - Users Mailing List
sr-users@lists.sip-router.org
Cc
yep, i do. I have the following entries in my config:
modparam(xlog, log_facility, LOG_DAEMON)
and closer to the top of the file i have
log_facility=LOG_LOCAL0
From: Corey Edwards ten...@zmonkey.org
To: mark li limar...@yahoo.com; Kamailio (SER) - Users
is there a way to add debug statements in kamailio.cfg? I'd like to be able to
dump some of the variables and also see which path the calls are taking.
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config parameter to something higher than 2. But
be warned that even 3 is very verbose. Start there:
debug=3
On 26 March 2014 15:53:31 GMT-04:00, mark li limar...@yahoo.com wrote:
is there a way to add debug statements in kamailio.cfg? I'd like to be able to
dump some of the variables and also
Hi there. I'm a noobie to Kamailio and Freeswitch... but I'm trying to follow
the article located here:
http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
I've tried to add all the sections marked with WITH_FREESWITCH in the sample
config in the article into my own
mohq_process() to send
inbound calls to a suitably defined queue, based on the incoming URI.
My question is what is the process for managing/automating the call to
mohq_retrieve() to send calls to agents? Do I need to track agents with XMPP
presence, etc?
Thanks,
Mark Hall
the auth is correct.
Is this doable?
If so, how would I go about setting it up?
It looks like Kamailio should be able to do just about anything, but I
don't know where to start.
Thanks.
Mark II
--
Mark D. Montgomery II
http://www.techiem2.net
biniUjaFZDVzD.bin
Description: PGP Public Key
We are using WITH_DEBUG, so we did see the cfgtraces, but they are
wholly incomprehensible to us, I'm sorry to say.
On Wed 04 Dec 2013 01:11:26 PM CET, Daniel-Constantin Mierla wrote:
On 12/3/13 5:17 PM, Mark Zeman wrote:
The SIP packets are fine - we are able to establish SIP session
and IPv6-IPv6! IPv4-IPv6 we
get a proper connection,
secured with SRTP, but no audio. Looking at the network, RTP packets go
from the caller to the server,
but nothing leaves the server and no RTP packets go from callee to server.
Do you have any idea how to fix this?
Cheers,
Mark
Dec 2013 03:27:55 PM CET, Klaus Darilion wrote:
On 03.12.2013 14:23, Mark Zeman wrote:
Hello all,
The subject says most of it, I think.
We set up our Kamailio and RTPProxy according to
http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
with the addition of an alias (siplab.ch
is well.
On 12/03/2013 05:17 PM, Mark Zeman wrote:
The SIP packets are fine - we are able to establish SIP session with a
secured control channel as easily as with an unsecured one.
As long as the RTPProxy doesn't get involved (i.e. as long as it's
IPvX-IPvX and not IPv4-IPv6) everything
/controller.php 6
Please ask system administrator for help...
Report to admin Show Error
Please wait a while ...
-
Any suggestions on what might be causing this?
I'm not sure where to start looking.
Thanks.
Mark II
--
Mark D. Montgomery II
http://www.techiem2.net
bin7HFP7pjUoc.bin
Hello everyone,
We're thinking of setting up conference call capability for our Kamailio
server, but we can't find any tutorials or documentation.
Could someone point us in the right direction or maybe even tell us how
to do it?
Thanks!
___
SIP
understand.
Does anyone know why there would be this delay?
Attached is our config, if that is of any help in fixing this delay.
Cheers,
Mark Zeman
#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v4.0 - default configuration script
# - web: http://www.kamailio.org
# - git: http://sip-router.org
actually
login and start using it?
Thanks.
Mark II
--
Mark D. Montgomery II
http://www.techiem2.net
binIxgYPDOM9V.bin
Description: PGP Public Key
pgpm1vMl0EbYD.pgp
Description: PGP Digital Signature
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SIP Express Router (SER) and Kamailio
rtpproxy instance to direct the audio.
Any comments or alternate solutions/suggestions would be of interest.
Many thanks,
Mark
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can't say if
everything else is working as expected :-)
Anyone have any thoughts?
Cheers
Mark
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instances, but was wondering
if this is the best solution.
Thanks in advance.
Mark
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to achieve is that the redirect is followed whilst
the other available phone carries on ringing.
I've tried with failure_reply_mode as the default 0 and as 3, both do the same
thing.
Currently working with kamailio version 3.4.0-dev6
Can anyone spot what I'm doing wrong ?
Thanks
Mark
to be the address of the Kamailio server?
I cannot see where to correct this without breaking tables the server needs to
know how to contact the phone when it needs to.
Has anyone seen this before?
Cheers
Mark
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SIP Express Router (SER) and Kamailio (OpenSER
it looks like presence_dialoginfo should be doing this but isn't.
What have I missed?
Cheers
Mark
--
Mark Boyce
On 27 Nov 2012, at 19:44, Andrew Mortensen wrote:
I recently added an SCA module to the project. If you're willing to try the
master branch, I'd appreciate hearing how things work
need to drop in the INVITE logic to trigger the notify?
At the moment I'm loading presence.so and presence_dialoginfo.so modules with a
config which is pretty much the same as the shipped sample config.
Cheers
Mark
--
Mark Boyce
On 29 Nov 2012, at 19:09, Ovidiu Sas wrote:
Do you see
Hi Ovidiu
Think I've cracked it. Thanks for your help!
Now to see if I can find a way to set the status when a phone subscribes. At
the moment it only updates when a new call is made.
Cheers
Mark
On 29 Nov 2012, at 19:57, Ovidiu Sas wrote:
You need to load pua_dialoginfo and configure
on OpenSIPs
http://www.opensips.org/html/docs/modules/1.8.x/presence_callinfo.html
Thanks!
Mark
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Hi Andrew
Thanks for that, I'll give it a go.
Would the module work with stable branch or do I need to compile up master?
Best regards
Mark
--
Mark Boyce
On 27 Nov 2012, at 19:44, Andrew Mortensen admor...@isc.upenn.edu wrote:
I recently added an SCA module to the project. If you're
hello
i want to active international call in my account for important .
radioyork2
thedept
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Hi Anton,
Previously I used the following.
sipsak
http://sipsak.org/
or
sipp
http://sipp.sourceforge.net/
Regards,
Mark
On Mon, Aug 13, 2012 at 3:41 PM, Anton Kvashenkin
anton.juga...@gmail.comwrote:
Hello, guys.
What is the best tool for crafting sip messages for testing purpose
it.
Thank you in Advance.
Regards,
Mark Anthony C. Delfin
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Hello list,
I am not sure if this has been ask before but is there a way that Kamailio
can sip received SIP method MESSAGES and then call a URL (Kannel URL).
Thank you.
Regards,
Mark Anthony C. Delfin
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Sameer,
Kamailio is not a replacement for Asterisk. We, and many others on
this list, use Kamilio+Asterisk+AGI to accomplish what we want.
Mark
On Fri, Mar 18, 2011 at 9:07 AM, Alex Balashov
abalas...@evaristesys.com wrote:
Sameer,
I suppose the Kamailio facility most analogical to your
This might also be of use if bandwidth is an issue:
http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/
Rgds,
Mark
On Thu, Nov 18, 2010 at 1:57 PM, marius zbihlei marius.zbih...@1and1.rowrote:
On 11/18/2010 03:59 PM, Fred Posner wrote:
On Nov 18, 2010, at 8:49 AM, marius
Dear Daniel,
With reference to your comment (in italics below) and request for a
documentation marshal, I humbly apply to be given the honor of contributing
to that effect.
We would need a documentation marshall that should update the docs as
something related is discussed on the
Dear Sir,
I just wanted to inform the good people on this forum that i had corrected the
problem with the Config file, and the server is running.
I noticed on going over the file again that i Fat fingered some keys ;)
Sorry for any inconvenience my post may have caused anyone.
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