Displaying 20 results from an estimated 3000 matches similar to: "SER + Asterisk"
2008 Nov 05
1
SER/Asterisk interworking mailing list.
Greetings,
As a developer and consultant who spends considerable time on projects
involving the fusion of Asterisk and products derived from the SER
ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have
found that there is a great volume of interest in this topic on the
mailing lists associated with all communities involved, but a
comparative lack of focus that results in
2008 Dec 13
3
SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
One of the above is frequently used to front-end Asterisk.
I used OpenSER to front-end a farm of Asterisk servers and was very happy
with it. The ability to take a box out of service or to route a specific
DNIS to a box for testing rocks.
Since OpenSER has died (I don't care about the
politics/personalities/trademarks), Kamailio and OpenSIPS have risen from
the ashes. What are you using?
2008 Sep 09
2
SIP to IAX?
Hi all!
I am looking for some software that would work as a proxy between a SIP
device (SIP phones and ATA boxes) and the Asterisk system running IAX. The
reason is that I can only get IAX trow the firewalls, and can't change the
settings.
One solution I am using are getting several Asterisk system communicate trow
IAX bout in this case would I rater have every persons computer act as a
proxy
2008 Nov 10
6
changing the size of voice packets
Dear,
is any way to change , the size of voice packets?
I want to increase the quality by decreasing the size of each packets, because of bandwidth failure.
?
thanks in advance
Mani
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2011 Mar 04
3
OT: OpenSIPS vs Kamailio -- which do you use and why?
I'm starting a new project similar to a previous project where I used
OpenSER to front a bunch of Asterisk servers.
Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely
candidates.
I'm leaning towards OpenSIPS because it's in EPEL so I can install it with
yum. Also, because I think the name sounds more 'professional' when
discussing architecture with clients :)
2009 Feb 11
2
OPTIONS packets
Hi all,
I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is
OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy
is not replying back...The issue is the UNKNOWN username that reside in the
OPTIONS packet as you can see in the captured packets as you can see below:
1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060
2. OPTIONS sip:OPENSIPS_IP
2009 Jan 19
3
[somewhat OT] seeking ideas/input for my thesis
Hello VoIP guys
Sorry for being somewhat off-topic. At the moment I am studying
informatics in the seventh semester and I need to start thinking about
my thesis. As I am very interested in VoIP technologies I thought about
picking this as my main topic. So far I have only little experience in
this area. I have been fiddling around with siproxd and pfSense and have
red the one or the other packet
2009 Oct 19
3
delay in processing dtmf
Hi,
I'm new to this list
I'm developing asterisk application where users can call and control volume
up and down in music player.
Problem I'm getting is if users press 222228 in fast speed, system will
process all those 2s and then process 8, so there is few seconds ( around
4-5) processing key press 8 , therefore users will feel unresponsiveness in
system.(in other words users will
2008 Nov 01
1
VoIP traffic shaping
This was so interesting I had to move it to its own thread!
Is anyone using this script? How does it perform compared to the older
WonderShaper script?
-M-
==================
Thanks Kristian I will checkout the new script and see how it goes!
Jonn
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at
2008 Dec 31
2
Friday VUC 12 Noon ET with Kristian Kielhofner: Identifying Asterisk Quality Issues
Happy New Year in advance by a few ticks for the northern hemisphere.
Here's the first topic and guest for 2009:
In any voice path there are several potential sources of quality
problems, ranging from
echo to voice dropouts and everything in between. With VoIP systems
the potential for
quality problems increases dramatically, often times making it very difficult to
identify the source of
2008 Nov 10
3
directrtpsetup without reinvite
Hi,
I want to be able to bridge two sip channels using direct RTP
between my endpoints (Audio IP : not local) but without
using reinvites. So I set up my asterisk sip endpoints as follows:
[test1]
type=friend
host=dynamic
username=test1
dtmfmode=info
context=test_rtp
allow=all
canreinvite=no
directrtpsetup=yes
[test2]
type=friend
host=dynamic
username=test2
dtmfmode=info
context=test_rtp
2008 Sep 19
2
Specific SIP answers on incoming calls?
Hi,
when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong
number" to unwelcome callers.
Meanwhile, I am only using SIP providers (no PSTN lines any more) and I
would like to do similar, i.e. send specific SIP headers. Besides "wrong
number", I would especially like to send 302 temp moved with a specified
address to deflect certain calls.
Is there any way to
2009 Feb 25
2
SheevaPlug Development Kit
Hello everyone,
I just ordered one of these:
http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp
Just over $110 with shipping but they are expecting the price to
come down quite a bit:
- 1.2Ghz ARM5
- 512MB RAM
- Multiple flash storage options
- Gigabit ethernet
- USB 2.0
- 5 watt power usage
They probably won't be shipping until late March but I
2009 Jan 30
3
looking for a link or pdf ot something about opensip/openser and load balancing
hi
i need a link or something about asterisk load balancing i cant find any, i
only found a paragraf in an email
anything wiil be wolcome
thanks!
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
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2012 Jan 03
4
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Hi,
Please help me understand the following applications and what are its
advantages if we compare between each of them.
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Regards,
Kaushal
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2016 Jul 05
2
OpenSIPS or Kamailio based fronting for Asterisk?
Hello,
I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so
far my biggest issue is the complete lack of quick-start-like documentation
for either. Is there any place I can get a very simple HA configuration
(telling me where the config files are, for starters, is a good thing) for
OpenSIPS or Kamailio with the following features:
(a) Support an arbitrarily large number of
2010 May 17
1
R: new way of asterisk and kamailio(openser) realtime integration
Works for me....
Thanks,
Hristo Benev
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru Oniciuc
Sent: Monday, May 17, 2010 6:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration
2010 May 17
1
new way of asterisk and kamailio (openser) realtime integration
Hello,
I put together a new tutorial about asterisk realtime integration with
kamailio (openser). This time the database used is the one of asterisk,
also call routing logic is controlled by asterisk, here is the link:
http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb
Practically is an easier way to scale starting from existing asterisk
installations.
The other
2009 Oct 15
3
DS3 capacity calls using asterisk
Hi All,
We are trying to implement a DS3 capacity calls (672 concurrent calls) using
asterisk server. I wanted to ask are there any compatible DS3 cards with
asterisk? I tried searching a lot but could find DS3000P from digium but
unable to get this product. Does anybody have any idea of having any DS3
card in asterisk box so as to handle around 600 calls?
Thanks
Sandesh
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2009 Jul 15
2
How to ask questions the smart way
Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's
classic "How to Ask Questions the Smart Way" to the OpenSIPS-users
mailing list[1], I'm going to repost it here:
http://www.catb.org/~esr/faqs/smart-questions.html
As Adrian said, "This a good read for those who show up on mailing lists
without any guidance about how to ask the right