Displaying 20 results from an estimated 10000 matches similar to: "Question about g729"
2008 Jan 14
2
G.729 pre-compiled binaries and Asterisk 1.2.x.
Asterisk 1.2.24 seems to crash repeatedly under any substantial call load
(and sometimes without a substantial call load - just one SIP leg is
enough to do it) when using the G.729 pre-compiled binaries from:
http://asterisk.hosting.lv/
As per:
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
Time to crash is variable, but seems to require at least an hour of
production performance
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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2009 May 18
7
callcenter / dialer / predictive dialer / vicidial program is now open
This is a global message to all to announce our callcenter / dialer /
predictive dialer / vicidial program is now open.
Codecs: G711, GSM, G729, G723
Protocols: SIP
Duration Rate : 30/6 (6/6 with monthly minutes over 100,000)
Channels : 100 to start with , more on demand.
We are predictive dialer friendly , your account will not be shut off.
Contact us to do a test run.
Mike
2007 Jul 03
5
Determining the used codec for the IP Trunk (SIP Trunk)
Hi List;
Where I determine the codec to be used for the SIP
Trunk (between Asterik and another SIP softswitch)?
Regards
Bilal
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2011 May 05
1
asterisk for g729 to g711
Hi,
Does anyone know if Asterisk is a good tool to be used for a large quantity
of g711 and g729 transcoding?
What is the best alternative for that?
--
Woody Dickson
woodydickson at gmail.com <woody.dickson at gmail.com>
US and Worldwide Termination
============ Contact me for the following offering ============
USA Onnet - 0.0049/min
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USA Mobile starting
2011 Jun 29
1
No audio format found to offer.
This *should* be something that's easy to fix, but apparently I'm not
doing something right.
Our SIP long distance provider is telling us to only use formats G.723
and G.729, so I've set up their trunk configuration in sip.conf as such:
[t564]
type=friend
host=XXX.XX.56.4
context=default
disallow=all
allow=g723
allow=g729
However, the Dial application gives the following error:
2008 Oct 27
11
Fring: Open VPN client to be installed on the mobile, which mobile?
Hi All;
I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile:
Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server?
Regards
Bilal
2008 Oct 29
4
Dimensioning a telephony system based on openser!
Hi,
I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk
1.4 + CDRTool with freeradius telephony system.
Asterisk is used only for voice mail and redirectioning calls.
Every calls should pass through mediaproxy so that i can account them.
The goal was to create a simple prototype of what could be a VoIP
provider.
Now i need to dimensioning this system to work
2007 Jul 30
5
Silly MeetMe() question.
I've got the ztdummy kernel module loaded and seem to have all the desired
prerequisites in place, but Asterisk never seems to compile with MeetMe()
application support enabled, nor does there appear to be a module I am
failing to load that would contain this application.
Is there something really obvious I am missing?
Thanks,
--
Alex Balashov
Evariste Systems
Web :
2007 Aug 03
4
PRI - DS3 Calls Dropped
I have a customer installation with an Adtran DS3 mux. The DS1's go into my Asterisk servers that run IVR/Call recorders. The DS3 provider is Qwest, and they tell me that they routinely drop the DS3 service to redundant back-up's and that this is a common practice that happens thousands of times to DS3 lines daily across the US without any service interruptions. They say that the
2007 Jun 22
10
inband DTMF for g729
Does anybody know why Asterisk does not support inband DTMF for G.729?
Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system.
Any suggestion to solve this problem?
Gary
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2009 Nov 02
5
Forward DID to another server
hello all,
i have 2 asterisk boxes on that 1 have public IP Address and another is only
have local IP address
now on public IP there are some 7 DID forwarded , now i want to forward 3
DID out of 7 DID to
local machine we called server B , I know there are DIal , and Switch
statement in asterisk ,
but is there any other convenient way to do this. because if call ratio is
high then my call legs
2011 Nov 27
6
Does Asterisk alter the Headers of INVITE Message
Hi all,
I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there
So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message?
Thank u
2008 Jul 19
2
OT Astricon/Digium Beach Ball Mailing
Just an FYI for Digium. I received a mailing today from you guys
which was nice. The price of mailing was ~$1.60 and inside was an
inflatable beach ball.
Cool, but I tried to blow up the beach ball and the the seam where the
part opens to inflate the ball was not connected to the ball
whatsoever, so it went right in the trash.
I wonder if the sick heat had anything to do with it, was mine just
2008 Dec 01
3
OT: What do you guys think of this?
http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/
FUD? Interesting? Boring? New news? Old news?
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
2012 Apr 27
2
Flashphoner
Really? Me?
Oh Pavel! I would be inestimably honoured.
On 04/27/2012 01:55 AM, Pavel Ismailov wrote:
> Hello!
>
> My name is Pavel Ismailov
> and I`m CEO of www.flashphoner.com project.
>
> We noticed that you quite active in Asterisk-user
> mail list, and would like to offer you buy signature
> in your messages for some monthly price.
>
> Is it interested for
2007 Nov 24
3
Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.
I made a little write-up that attempts to synthesise a lot of the
information out there about how to get HylaFAX working with Asterisk
by way of IAXmodem for inbound faxing:
http://blog.evaristesys.com/?p=24
Of course, there are bound to be some things I've left out or are grossly
in need of correction. So, before I link it off the voip-wiki I am
extremely eager to solicit the input of
2008 Aug 21
1
DSS1 vs SS7
Hi,
I am requesting for a E1 connection from my telco. They are asking if I
want DSS1 or SS7, and I am stuck here. Could someone tell me the difference
between the two? How should I decide which one to use?
Thanks in advance for your help.
Mark
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2008 Feb 04
6
transcoder
Dears
Any one knows a standalone voip transcoder software name,not an ip pbx.
What I want is to transcode the incoming sip calls from g711 to g723 or
ilbc or g729 ..... and forward it to a media gateway ..
Regards
Khaled chehab
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2007 Sep 18
6
Limiting Simultaneous calls
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
Thanks
Jim