SIP is a XML-like control channel and is used to negotiate a separate RTP
channel which carries the audio. It is complicated to set-up in cases of
firewalls and NAT, but is an open standard.
IAX2 is a candidate open standard and merges all traffic onto a single UDP
stream - control and audio
I have had the same problem when calling across Asterisk from Diax to a SIP
phone. If Asterisk Answers the call before the Dial to the SIP phone
there is no delay. Otherwise there is a 10-20 second delay in the Voice
path!
Peter
-Original Message-
From: Dan [mailto:[EMAIL PROTECTED]
GSM Codec is 13k plus overhead. That may work?
Peter
-Original Message-
From: Bilal Ghayad [mailto:[EMAIL PROTECTED]
Sent: 15 January 2005 07:07
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] DIAX
Dear Dan;
Thanks alot for your kindly reply.
Well, what u advise us to
Hi
This is my script for my local forecast for SE England. I have had
problems getting festival to work integrated so I have cron run this
script every 3 hours and use Playback to play it in Asterisk:
Script
--
#!/bin/sh
cd /var/lib/asterisk/sounds
curl
There are issues with Asterisk chan_sip. Have a look at bug 759 at
bugs.digium.com. Comments in the feature report and source code like
those below probably go a way to explain your problems. I don't know how
much of this test version has been back-ported to chan_sip, however the
chan_sip2.c
I had a compile problem with the CVS I downloaded on 21 June.
I have a Debian box with 2.4.18 kernel (version needed for support of
Conexant ADSL). There is a difficulty with Zaptel build regarding HDLC
detection. It tries to build it in and then results in unresolved kernel
symbols and fails to
I get a problem with what appears to be a slow oscillation on the line if
the rxgain + txgain adds up to more than -1db. If I use rxgain=-1.0 and
txgain=0.0, it doesn't oscillate but the levels are far too low. The card is
an X100P.
The oscillation (even on the standard built-in Asterisk echo
BT do occasionally tweak up line gain a bit if you keep complaining that you
have a modem and are getting a very slow speed. I have had a 40k V90 come up
to 48k after this was done on my line at home (System X switch).
You have to get a sympathetic engineer though - frequently they will tell
you
Has anyone seen this problem before?
I have a server with a single X100P card. The audio level is a low, but if I
raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo
test. Not at a high frequency but with a noise that is best described as a
steam engine starting up. It
, Whisker, Peter
[EMAIL PROTECTED] wrote:
I have tried the latest CVS Head with echotraining=800 set and also
complied
with the aggressive echo cancelling, but nothing seems to help.
Ideas welcome!
Many thanks
Peter Whisker
This e-mail and any attachment is for authorised use
any rx/txgain value. As soon as the call utilizes two FXO
card at the same time, the steam engine sound occurs.
On Mon, 2004-06-28 at 16:26 +0100, Whisker, Peter wrote:
Has anyone seen this problem before?
I have a server with a single X100P card. The audio level is a low, but if
I
raise
Maybe it is trying to say i as a digit?
You could have an [invalid] context with
[invalid]
exten = _.,1,Saydigits(${EXTEN})
and then include it at the very end of the [default] context (or wherever
you want to use it). That would then pick up anything that drops through. If
you do it any other
I get a compile warning when building zaptel (current CVS head) against
2.4.18 kernel (Debian stable dist)
zaptel.c: In function `zt_net_close':
zaptel.c:1238: warning: implicit declaration of function `hdlc_close'
It completes but fails to install with modprobe finding unresolved
references.
-28 at 16:26 +0100, Whisker, Peter wrote:
Has anyone seen this problem before?
I have a server with a single X100P card. The audio level is a low, but if
I
raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an
echo
test. Not at a high frequency but with a noise that is best
I have two Asterisk servers interconnected with IAX (non-trunk). I place a
call on Server B (using DIAX) which goes to an extension on Server A and
terminates with a Dial to a local SIP phone (Sipura SPA 2200).
The SIP phone rings immediately but when it is answered there is a delay of
about
I picked up a Sipura SPA-2000 (new) on e-Bay for ~£70. The voice quality is
excellent.
Peter
-Original Message-
From: Mike Dent [mailto:[EMAIL PROTECTED]
Sent: 29 November 2004 14:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] UK available
Partly is is down to the fact that G.711u (mu-law) is primarily used in the
USA and G.711a (a-law) is used in Europe.
Like you, I am not sure if the exact differences - they have the same
bitrate and audio, although there are minor differences in the format.
Peter
-Original Message-
For
info
The
new chan_sip2.c and recent CVS (yesterday) fix this and I can now use Asterisk
to make calls on the sip.btcommunicator.bt.net service. If anyone wants help
withthe settings, e-mail me off list.
:)
Peter
-Original Message-From: Whisker, Peter
[mailto:[EMAIL
above.
Regards
Peter
-Original Message-
From: Robert Boardman [mailto:[EMAIL PROTECTED]
Sent: 09 October 2004 21:40
To: Whisker, Peter
Subject: bt communicator`
Hi Peter
I have been following your post but didn't see the other emails about
getting it working until now!!
Could you please
compared all combinations
of MD5sum with the ethereal trace and cannot see it any where?
Still cannot register, any advice would be greatly appreciated
Regards
Robb
Whisker, Peter wrote:
Hi Robert;
First, you have to use the SIP2 channel code (chan_sip2.c) from
http://bugs.digium.com
Same for me. After a few minutes the program crashes.
Any chance of support for ULAW / ALAW which is mandatory for FWD IAX?
Thanks
Peter
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: 14 October 2004 09:33
To: Asterisk Users Mailing List - Non-Commercial
You would need a TCP version of IAX to use SSH as I don't think it supports
UDP.
Asterisk does work (tunelling IAX) through Zebedee (an SSH-like TCP UDP
tunnel).
Peter
-Original Message-
From: Tom Neville [mailto:[EMAIL PROTECTED]
Sent: 13 October 2004 16:55
To: Asterisk Users Mailing
Adam
On UK keyboards ,I have to type a £ to get a # into Firefly. The proper
# key does nothing. If you are updating the code, perhaps you might look
at this?
Many thanks
Peter
-Original Message-
From: Adam Hart [mailto:[EMAIL PROTECTED]
Sent: 16 October 2004 07:46
To: Asterisk Users
The SIP client in Windows Messenger 5.0 seems to work fine with Asterisk
though.
Peter
-Original Message-
From: Robert Rozman [mailto:[EMAIL PROTECTED]
Sent: 11 October 2004 22:08
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
It looks for tones (currently hardwired as US). I have updated to include UK
tones but is hard to get it to reliably recognise. For example the tones in
the switch here at work are 5-10% off frequency. Correcting for this, and
doing a lot of fiddling it did recognise the tones but was unreliable.
I have an * switch at home and one in the office. Both similar new CVS head
versions and both with chan_sip2 built in:
Asterisk CVS-HEAD-10/12/04-17:43:26
Asterisk CVS-HEAD-10/13/04-12:53:52
One is on a T1 connection and the other is on 576k/288k ADSL. The Ping time
is about 30ms between the two
[sorry about previous mis-post]
I have an * switch at home and one in the office. Both similar new CVS head
versions and both with chan_sip2 built in:
Asterisk CVS-HEAD-10/12/04-17:43:26
Asterisk CVS-HEAD-10/13/04-12:53:52
One is on a T1 connection and the other is on 576k/288k ADSL. The Ping
Oops. Sorry about this post. Pressed the send button my mistake!
-Original Message-
From: Whisker, Peter [mailto:[EMAIL PROTECTED]
Sent: 02 November 2004 14:18
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] OpenSource Proxies ?.
I have
I am a
little confused aboutvoice data transcodingin Asterisk. I can make a
call between twou-law-only phones over an IAX GSM-codec link and the two
Asterisk servers handle thetranscodingulaw-GSM...GSM-ulawfine.
However, over a SIP channel, this doesn't seem to work. Asterisk appears to be
This is my codec translation timing table (500MHz PIII).
The fastest codec is G.711 (ulaw/alaw) which is uncompressed, next is GSM.
The slinr-codec row gives the amount of time (and probably relates to
processing power) in coding and the codec-slinr columns are the decoding
time.
Speex is
I used DEC's EDT for almost 20 years on PDP-11 and find jed with the EDT
interface useful!
You can't teach an old dog new tricks!
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shadowym
Sent: 16 October 2007 17:10
To: 'Asterisk Users Mailing List
Can someone please tell me if it's possible to select the G726 codec
bandwidth for an IAX trunk between two Asterisk 1.2.5 servers?
I can select disallow=all / allow=g726 but I think it defaults to the
g726-32 variant.
Is there any way of forcing Asterisk to use g726-24 for such a trunk
On Wed, 18 Apr 2007, Joseph wrote:
Are there any cell phone (gadgets) that can be connected to standard
switch phone network? (ability to check email would be a plus).
Digium adapter S101i can be connected to any network and it allow a
standard phone to act as your local extension over
I have been looking at the medium-rate codecs in Asterisk - ADPCM and
G.726. Both of these are adaptive PCM codecs - the G.726 one is a little
more expensive in processing power, however both are 32k bit-rate.
I am experiencing problems using G.726 where the audio level is high. It
produces loud
-EFR mobile phone. And the rest
are worse or like G.729 or Speex, eat processor power. I have G.723 and
G.729 compiled from the Intel distro.
Peter
Steve Underwood wrote:
Whisker, Peter wrote:
I have been looking at the medium-rate codecs in Asterisk - ADPCM and
G.726. Both
I managed to get it to work and make a test call from Asterisk to Skype.
Pity it's not implemented on Linux and needs Skype to be running on the
PC also.
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Whitten
Sent: 05 April 2006 16:18
To:
Dante's DIAX is pretty good IMHO.
Peter
Hector medina wrote:
can anyone recomend a good iax softphone??
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I have tried to get this working, but can not get it to authorise:
I created my Communicator logon from a Yahoo account (not a btinternet
account). Assuming my Yahoo username is username, BT Communicator
software logs on to the SIP proxy as username[EMAIL PROTECTED]
according to the trace which
=btinternet.com
fromuser=username.brz
md5secret=md5hash
host=sip.btcommunicator.bt.net
;/etc/asterisk# echo -n
[EMAIL PROTECTED]:btinternet.com:password | md5sum
;md5hash -
Peter
-Original Message-
From: Whisker, Peter [mailto:[EMAIL PROTECTED]
Sent: 06 September 2004 09:18
To: 'Asterisk Users
I am
getting this also.
I am
trying to get Asterisk to talk similarly to BT Communicator to the BT server. I
can register but then the INVITE fails.
BT are
mixed up with theirdomains (in fact in the INVITE their software has a To:
header withnumber@domain1 and an auth URI referencing
I have an Asterisk servers (recent SVN version 1.2) and two Sipura ATAs
(one 2000 and one 1001).
I have Three-way Conf Serv and Three-way Call Serv enabled on both ATAs.
When I make a SIP call from phone 1 to phone 2 on my Asterisk box, it
works fine, then when I press the hookflash on phone 1,
Where bandwidth is not an issue but good call quality is, is there any
theoretical quality improvement to be had by using slin as the codec
over an inter-Asterisk IAX trunk rather than a-law (or u-law in the US).
Does anyone know what the slin bandwidth is (compared to 64 kbps a-law).
Thanks
Partially answering my own question, it looks like slin is a 128 kbps
codec.
Peter
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Whisker,
Peter
Sent: 05 December 2007 16:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
It is possible to run openVPN in TCP mode over an SSH tunnel. Don't turn
compression on on both though - I'd just switch it on the openVPN if you
have to.
You will probably find the speech is rather choppy due to the delays and
fragmentation, but I have done this.
Peter
-Original
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