Displaying 20 results from an estimated 40000 matches similar to: "Help: codecs and bandwidth"
2014 Dec 31
0
Operating with different codecs - can't native bridge...
When I try to dial out I get an error:
Operating with different codecs [0x2 (gsm)] [0x4 (ulaw)] , can't native bridge..
Here are the details:
-- Accepting AUTHENTICATED call from 66.18.210.217:
> requested format = gsm,
> requested prefs = (),
> actual format = gsm,
> host prefs = (gsm|ilbc|ulaw|alaw|speex),
> priority = mine
--
2003 Jun 28
1
IAX2 trunking: codec bandwidth comparison notes and results
2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com)
Purpose:
-------------
To obtain a better chart of actual bandwidth usage per codec as
seen "on-the-wire" when using IAX2 trunking between two Asterisk
telephony servers.
Discussion:
-------------
Past threads on the asterisk-dev and asterisk-users lists have
indicated that the optimal way to save bandwidth on
2005 Jan 19
1
Re: Asterisk bandwidth tuning?
Well, I don't know how to tune it more, it connects at about that rate in a mediocre
rural landline.
ILBC uses samples of 30ms, so if you set the trunkfreq set to 20 you will be using more
of the necesary scarce bandwidth AND dropping sample info in each frame, thus making
audio choppy and unclear.
Make shure to disallow all codecs and then allow only ILBC or lpc10 (search for it in
2005 Jan 18
1
No compatible codecs
Original Post
----------------
I have an Asterisk related problem with mutualphone.
I can connect to any number with any softphone that I am using (iaxcomm,
SJPhone, and a few others).
Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to
mutualphone destinations. Other destinations go fine.
A working phone call (e.g. from iaxcomm) gives the following on the
console:
--
2005 Feb 11
2
Codec Issue on IAX trunk?
Hi All -
Well, after happily existing in a one office environment with asterisk
for a few months, I've now decided to start adding in our other offices
with their own * boxes and IAX connections (over VPN). Unfortunately,
I'm an idiot and I can't get it to work. I'm having some kind of
problem with codecs, I guess, but I don't understand what or why. When
trying to use
2004 Dec 14
0
Codec "Uknown" with IAX connection
I am having some problems getting TelIax service to work with *. Outbound
calls work just fine. When I try an inbound call the phone rings and there
is no audio. Upon further investigation "iax2 show channels" indicates
that the codec is "unknown" The provider confirmed that they are set for
ulaw and so am I. Does anyone have an idea what could be causing the codecs
to
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)
--
2004 Jan 29
1
re: help with voicepulse connect IAX2
hello,
after playing with an asterisk configuration for voip for a few weeks i'm
trying to get outbound dialing with voicepulse going - i've cut down the
asterisk to a very minimal install (1 SIP client) to try to localize the
problem. The SIP client works fine (SIP and * on the same NAT) and could
access the demo from samples before i removed it, and can call itself - so
i am
2008 Nov 13
0
Problems with Licensed g729a codec from Digium
Firstly, I'm running Asterisk 1.4.4 on Solaris 10.
I have several different internal SIP phones all sharing a single IAX2
VoIP channel.
PHONES |------------- <SIP/uLAW> --------------| ASTERISK
|-------------- <IAX2/g729> ------------|VoIP/ISP
The g729 codec has been registered successfully and appears to be
detected by Asterisk
(NOTE: I have changed what I thought might have
2004 Sep 13
4
Unknown RTP codec 72 received
Hi all,
I get "Unknown RTP codec 72 received" message in console when call in
progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN
over voicepulse connect (IAX) and to FWD echo test (SIP). But this
message only with one SIP client, others (X-Lite too) not giving this
message. All X-Lite settings are identical. Asterisk is last cvs version
This what I see in console
2005 Mar 12
2
Unable to create channel of type 'IAX2'
Hi all,
I'm a newbie and I have a configuration problem with Asterisk.
Seems that I'm not able to call an outbound number. I'm quite sure that it
is a configuration problem, but I'm not able to find out where is the
mistake, even reading several docs to www.voip-info.org.
I do not have a good knowledge of Asterisk, I'm not very familiar with its
configuration and I've a
2004 Dec 12
1
Totally LOST with dialplan and Extensions.
Ok I have spent the last week working on getting my small PBX to work.
I will in the end only have 4 SIP extensions being either softphones of
IP phones. Currently only 1 SIP config for testing.
And at the this point it should be all fairly easy with all inbound and
outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via
IAX. Inbound does work in it's current basic state.
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
2004 Nov 18
3
iaxComm to iaxComm
Having some trouble with segfaults and sound quality all of a sudden (since
I recompiled from the latest source) when 2 iaxComm clients connect. First
off immediately after the server reports:
<>
<> -- Attempting native bridge of IAX2/4587@10.9.1.32:4569/1 and IAX2/4589/5
<>
<>One or both client may sometimes segfault. Additionally, when they do
get properly
connected,
2009 Aug 04
0
SIP server behind NAT
Hello.
I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage
to make outbound calls, but the communication drops off after 30 seconds
or so.
I'd really appreciate having some assistance from the mailing list on
this issue.
So, I'm having an Asterisk server behind a firewall and Zoiper
softphones on SIP connecting to Asterisk on the same local area network.
The
2005 Jan 25
1
Codec mismatch between SIP (BT) and IAX Phone
Hi,
I have strange problem. I have 1 SIP client (bt100) and 1 Iax2 client
(IAXPhone):
- when I call from Iax to SIP sound works
- when I call from Sip to Iax sound doesn't work, I get :
Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping
incompatible voice frame on IAX2/200/1 of format gsm since our native format
has changed to ulaw
Why is Asterisk not satisfied with gsm
2004 Apr 07
1
Out of trunk data space on call number 16386, dropping
Hi all,
We keep getting these and all the calls between these two asterisk boxes get
dropped. what is going on here, I have been trying to solve this problem on
my own but maybe I don't have the trunk setup right. also I have posed the
output of my full log of the machine with the zap interface, the other is
using ztdummy.
IAX.conf on machine 1:
[general]
port=5036
;iaxcompat=yes
2010 Feb 08
0
Help with iax.conf {tesco|freshtel} 1.6
I have something going on that I don't fully understand after a weekend
of looking for answers.
I have an iax account with Tesco that works flawlessly with the Zoiper
client - but is giving me trouble with inbound calls in Asterisk 1.6.
After some playing I have ended up with an iax.conf file that looks like
this:
[general]
calltokenoptional = 77.75.0.0/255.255.248.0
maxcallnumbers = 16382
2003 Oct 10
2
Actual audio bitrates
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Hash: SHA1
Hi,
I was just measuring the bitrates of a couple of codecs via iax. I'm getting
much higher numbers than expected, so maybe I'm doing something wrong?
Measured with iptraf, values displayed are:
codec: measured bitrate (bitrate according codec definition)
gsm: 52 kbps (13 kpbs)
alaw: 154 kbps (?)
speex: 57 kpbs (24 kpbs)
Seems a little
2005 Mar 18
0
IAX Peer/auth issues WAS: Netlogic inbound DID issue
Has something changed in the recent modifications to Asterisk that would
break dialing of the IAX peer? We're getting these authority failures
everywhere.
Everything is configured just the way it was half a year ago, this is
the message we're getting on the most recent vers of asterisk. Wiki says
nothing, nor does the ast-dev list..
-lost
Mar 18 12:55:23 NOTICE[3479]: chan_iax2.c:6545