Displaying 20 results from an estimated 10000 matches similar to: "Server Criteria"
2006 Oct 18
1
Server power indication
Hello list,
I'm currently looking into building a new Asterisk server, due to some codec problems i've got to transcode most of my channels between
Alaw -- G729. Is there any indication on how many channels you would be able to transcode on a certain platform?
I'm looking into dual Xeon or dual Opteron configurations, which of these platforms would perform better?
And how much power
2005 Jan 31
2
Trunked IAX or not
>> Has anyone benchmarked Asterisk on a dedicated single versus dual
>> processor machine?
>
> http://www.astertest.com/
>
> Cheers, Philipp
The test results that Philipp pointed out show some protocol
comparisons that include "iax2 trunking / alaw" and "iax2 / alaw" and
concludes that "IAX2 trunking is more than twice as fast as non
trunking
2009 May 06
2
Understanding Codecs
Hi,
I'm having problems with an asterisk server that's not offering Codecs
for ulaw and alaw as it should.
I've three servers in total: a1, a2 and "b"
A1 and A2 have pretty much the same config files, except IP address info
changes
Server B is configured to accept all inbound invites.
Calls from A1 to B, all work fine, and in a sip debug session I can see
2005 Jan 07
4
glm fit with no intercept
Dear R-help list members,
I am currently trying to fit a generalized linear model using a binomial
with the canonical link. The usual solution is to use the R function glm()
in the package "stats". However, I run into problem when I want to fit a
glm without an intercept. It is indicated that the solution is in changing
the function glm.fit (also in "stats"), by specifying
2004 Sep 10
2
Suggested Motherboard for TE410P
Hi all,
I'm looking for a new system which will use the TE410P. Originally I was
going to use a dual Athlon MP system, but my supplier tells me these are
being phased out now, and so will be difficult to find replacement parts
later.
So, I am looking for suggestions of suitable motherboards with 3.3V PCI
slots for the following CPU types (in order of my personal preference)
AMD Opteron 1xx
2009 Sep 09
1
CLI file convert from alaw to g729 with TC400B transcoding card results to an empty file
Good afternoon,
I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card.
The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty
file as you can see below...
CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw
2006 Jun 15
3
SIP codec preference order ineffective
Hi,
I set a preference order of the codecs to my sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls of not registered phones
disallow = all
allow = g729
allow = g723
allow = alaw
allow = ulaw
Connected a 'Sipura SPA' sip phone to asterisk with g729 as its preferred codec.
Problem: asterisk cannot make
2010 Mar 23
5
G.711a or G.711u ???
Dear all, I have an Asterisk SIP server in a LAN environment and I want your
opinion in order to decide the use of an audio codec:
What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip
calls ???
Thank you !!!
Alejandro
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2008 May 12
2
Which sound file formats?
I've got the text files created -- thanks to Russell Bryant -- for
re-building the core and extra sounds using another voice but I'm not
sure which formats to actually build.
This will be a small/personal system using Vitelity.net so will only
have SIP connections.
The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729,
.gsm, .ulaw, and .wav.
What are the minimal
2008 Nov 11
3
Use the NEW ulaw/alaw codecs (slower, but cleaner)
In Asterisk 1.6, there is an option to use the 'new g.711 algorithm'.
"Use the NEW ulaw/alaw codec's (slower, but cleaner)"
By slower does this mean more 'expensive', or does it instead mean that there will be more algorithmic latency? Both? Can anyone speak to the relative increases?
With regard to accuracy, can anyone speak to what kind of situation might
2007 Sep 26
2
My G729 problem re-visited
Ok, I built a test system to duplicate my problem and provide myself
a platform that I can mess around with to try and break any features.
My problem is G729 pass-through from a gateway to a phone. I think
I even have transcoding working, which makes me more confused on
what's wrong with my pass-through. It must be a configuration issue.
The basics...
*CLI> core show version
Asterisk
2006 Mar 21
3
Zap<-->IAX codec?
Hi,
at my Asterisk box, I have a few of IAX2 phones (configured with
alaw/ulaw/gsm codecs, in this order) and a PRI E1 line.
In iax.conf I hav:
disallow=all
allow=alaw
allow=ulaw
allow=gsm
During some incoming call, I read at console:
-- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack
-- Called 215
-- Call accepted by 10.97.1.7 (format ulaw)
--
2014 Mar 02
2
[LLVMdev] Stub LLVM backend wanted
I'm trying to port LLVM to a new architecture. I'm finding that the
initial bootstrapping stage of getting something which will build, even
if it doesn't work, is complex and rather disheartening --- there's this
huge cliff of difficulty in just getting all the boilerplate laid out
correctly, before getting to the fun stuff. The other backends are of
limited use here because, of
2008 Mar 28
3
Two phones fail to agree on codec, asterisk at fault?
Hi list,
I am faced by a situation where I am trying to make a softphone and
a Siemens C450IP talk to each other. Both are hooked up directly to
the same asterisk, in the same IP net.
- a softphone runs on 192.168.14.3
- the C450IP is at 192.168.14.30
- asterisk runs on the machine known as 192.168.14.1
I am running Asterisk 1.4.11, backported to Debian Etch by Xorcom.
If I set
2004 Sep 10
4
SIP on Handhelds
Does anyone know if SIP will/is support on handheld PCs such as the iPaq
or Axiom? With their integrated 802.11b and Bluetooth it seems like a
solution to provide a wireless based sip phone for any user would be
possible. Handoff between access points might be problematic but most
users I know would be using their PDA phone in an airport with free
wireless or at the local cafe, etc, etc...
Can
2006 Oct 25
2
Choice of soundfile format
Hello
What soundfile format, is the one that uses least transcoding during playback?
As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct?
Kind Regards
Jon Leren Sch?pzinsky
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2008 Aug 12
7
New Ogg Dirac mapping draft
David Flynn has proposed a new Ogg Dirac mapping. The draft is here:
http://davidf.woaf.net/dirac-mapping-ogg.pdf
This is a much bigger break from other codecs than my draft (at
http://wiki.xiph.org/index.php/OggDirac). We talked a bit about it on
IRC today. Below is my summary; hopefully David can correct anything
I got wrong or misleading. Comments?
There are two main differences
2020 Jun 05
2
Advanced Codec Negotiation: Need info and uses cases
Greetings All,
We've been working hard on new codec negotiation stuff for Asterisk 18 and
we've got some stuff to run by you. It's a lot so please read carefully.
To give you some idea of just how difficult a job this is, a simple call
from Alice to Bob currently causes 8 attempts to reconcile codecs between
them in app_dial, chan_pjsip, res_pjsip_session and res_pjsip_sdp_rtp. If
2007 Sep 19
18
sip.conf best practices?
All - I've been wrestling with how to best structure the sip device
accounts on a new asterisk server I'm deploying. All of the sip
devices (currently only Linksys SPA941s) will reside on the same
subnet as the server, and I have already set up a decent automatic
provisioning system for the phones. When the rollout is complete,
there will be about 100 SIP devices authenticating and
2019 Dec 03
4
Delay on speak with Asterisk
Hi list!
I'm using Asterisk 13.14.1 from Debian 9 repositories.
The provider is Deutsche Telekom und Messagenet (just for receive).
I can call and receive calls, but I have a little problem: there is a
"delay" of about 1-1,5 seconds between the time the voice is sent and
the time when the voice is received, so that it happens very often that
the peer does not get my voice and try