Displaying 20 results from an estimated 10000 matches similar to: "Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning"
2007 Nov 02
1
Jitterbuffer issues
2006 Jan 22
2
Disposition codes in CDR
Is there any way to have more specific disposition codes in the CDR?
Currently there are only 3 values: ANSWER, NO ANSWER, BUSY.
In this way, when i call a cell phone that is switched off i get "NO
ANSWER", while i would like to be able to log that the call is not
answered because "The customer you have dialed is unavailable at the
moment".
The same for "non
2006 Apr 12
1
iax2 show netstats
Hi guys,
i've been using iax2 show netstats and i wonder if someone could explain what all these means, just
in case i have them wrong. Because i am looking for something that tells me that there is delay ,
and/or packet loss.
-------- LOCAL --------------------- -------- REMOTE --------------------
Channel RTT Jit Del Lost % Drop OOO Kpkts
2005 Feb 12
2
Intermediary jitter buffering
Hello,
I understand that only the destination of a call should do jitter
buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no
transfers), PhoneA and PhoneB need to perform their own jitter buffering,
and Asterisk will just forward the frames, correct?
What happens if the peer does not support jitter buffering, but is
close by so there's no need for jitter buffering? My
2006 Feb 17
4
Bridged line appearance
So are there any plans for bridged line appearance support in Asterisk? The
new Linksys SPA9000 supports it. A lot of other VoIP systems from Nortel,
Sylantro etc. supposedly support it. Seems to me that Asterisk needs to
get on the bandwagon or be relegated to call centers, specialized voicemail
applications, and phone chat businesses. It's not needed for companies used
to PBX's but
2020 Mar 02
2
No CID between Asterisk using IAX trunk
Not these particular two servers.
On 02/03/20 12:16, Doug Lytle wrote:
>>>> I am trying to troubleshoot two Asterisk servers that have an IAX2
>>>> trunk between them.
> Carlos,
>
> Had caller-id ever worked between these two systems?
>
> Doug
>
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161
2006 Mar 21
1
Problem with chan_iax.c implimentation causesbad audio?
We have three remote call center Asterisk servers communicating with two
central Asterisk boxes over a private IP-VPN with QoS. All systems were
running Asterisk 1.0.7 communicating via IAX2 with little or no quality
issues at all.
Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific.
We tried with/without jitterbuffer. We messed with every jitterbuffer
parameter. We tried
2007 Apr 28
7
Two Connected Servers Sound Quailty
Ok this is my first post and I will try to keep it short.
I have searched everywhere and haven't found an answer to my question
I have two Trixbox servers that are connected over the Internet via an IAX2
connection. We are experiencing very poor sound quality. I have tried many
different codecs gsm, ilbc, g729, g711 and all seem to have the same
problem. (All though g729 seems to work the
2004 May 13
4
IAX Freeworld
I have looked all over the site(s) for help. But heres the problem. Im
missing something.
In coming works fine from FreeWorld via IAX. But when Dialing out i get:
May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I
don't know how to authenticate iaxtel to 65.39.205.121
my IAX.conf if as follows
[general]
port=5036
register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net
2006 Oct 24
2
IAX2 goes "one way audio" when lag gets bad
Hi,
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
The customer is connected via IAX2 to our softswitch.
On the customer's end I have the following config in iax.conf:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0 ; Address to bind to (all
2006 Mar 20
3
Problem with chan_iax.c implimentation causes bad audio?
I received an e-mail from a vendor who says:
"We have recently become aware of an issue in the chan_iax2
implementation of IAX2. This issue leads to degraded audio quality.
Due to this we are urging everyone to move to SIP."
I don't want to discount what this person is talling me, but I'm
curious to know why I would only be having issues connecting to his
servers, and also what
2006 Feb 07
1
asterisk to FWD
Hello all,
Here is my problem,
I try to place a call to FWD (free world dialup) trough my asterisk PBX.
my config is as follow:
extensions.conf
----------------
[internal]
exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD)
exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD
exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD
2004 Nov 17
3
Jitter buffer
Jean-Marc Valin wrote:
>>Heh. I guess after playing with different jitter buffers long enough,
>>I've realized that there's always situations that you haven't properly
>>accounted for when designing one.
>>
>>
>
>For example? :-)
>
>
I have a bunch of examples listed on the wiki page where I had written
initial specifications:
2003 Dec 18
1
Excessive VNAK's and jitter over IAX2
Howdy,
I recently saw something strange with a call between *'s over IAX2.
There are actually 3 *'s involved. The setup is like this:
SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over
Internet) ---------*2--------(GSM over Internet)
-----------*3--------(ulaw over LAN)------ SIP phone
Now what is shown below is the Asterisk in the middle, that is doing the
2007 Mar 13
1
IAX2 Question (Asterisk 1.4 tarball)
I've got IAX2 setup between two servers with this config:
I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is
192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls
from asteriskm to asterisk1 which will run an AGI IVR for the call.
Config is below, but my problem is that 90-95% of the time when I start
asterisk on the two servers I get the
2006 Mar 21
1
Problem with chan_iax.c implimentation causesbadaudio?
We upgraded all five servers to 1.2.4. We tried trunking/notrunking.
End users use an IAX2 softphone on their desktop PCs. Agents are VLANed
and all IAX2 traffic is QoS'd on all LAN and WAN legs. Calls flow from
the agents to the local Asterisk server as IAX2/ulaw. Then they went
over the IP-VPN as IAX2/g729 (we tried ilbc and straight ulaw as well).
Calls get to the PSTN from the
2006 May 03
2
New jitter.c, bug in speex_jitter_get?
> Perhaps, but then you need to assume that the jitterbuffer can just
> throw away the data, and that limits how you can use it. In object-
> oriented terms, you might want to pass objects to the JB, and then
> call a destructor on them. In C terms, you may want to allocate
> frames via malloc(), and then call free() on them later. You might
> want to pass in
2007 Apr 20
6
How can I improve call quality?
Hi All,
I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used
for PSTN calls via IAX2.
Our 'net link is a dedicated 2Mb fibre connection (of which we have ever
used 50% max bandwidth). We've no E1/T1 links, everything is IP based.
My boss complains that many of the calls he holds with others has a bad
quality. He also says that its not just him.
My iax.conf file
2009 Nov 21
3
Connect two Asterisk Server in IAX ?
Hi
My first post get no answer :=<, i post new with new elements.
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
on Srv1:
iax.conf:
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
2006 Nov 29
2
Trouble using 2 IAX2 DiDs provided by different ITSPs
Asterisk 1.2.7
Redhat 9
I have DiDs from two different ITSP both set up as IAX2. Each one
works when it's the only one in my iax.conf, but when I have them both
defined in iax.conf at the same time, only one will work. My iax.conf
is provided below.
Any ideas how to fix? I'd like to use both DiDs!
Thanks,
H
My iax.conf is below. When I dial the DiD provided by ITSP_B, the
other