asterisk users - Apr 2006 - Delayed voice for 10 secs

Hi List !!

I have a lot a questions about this incredible tool but short is my time to
learn it, so I apologize if my last question was too general. I got another
more especific trouble. I administrating an ISP and I have my Asterisk
installed on a server for testing my network performance. I followed the
quick-start tutorial provided by voip-info.org (which I think it's very
useful) and configured two SJphones as extensions. My network is HFC type
and the users surf the Internet by a Cable Modem (Motorola). When I tried
this 2 softphones, the voice was delayed for 10 secs aprox. and the next
warning is jumping on my screen:

WARNING[26673]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on
codec gsm. Use RFC2833

I know I must change codecs in order to get the voice more fluency but I
don't know yet if I have to configure it on the Asterisk server (on
sip.conf)
or somewhere else (on the SJphones). Can you give me some info about it? I
would appreciate a lot

Thanks

Carlos Bernat
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060419/26f1bf19/attachment.htm

Please post pertinent config files and a CLI output so the list can help
with the 10 sec delay

You set codec selection in SIP.conf. This selects preferred codec from
top to bottom as well as jitter buffer settings and the RTP timeout.

Sip.conf
disallow=all
allow=g729
allow=gsm
allow=ulaw
jitterbuffer=yes
;forcejitterbuffer=yes
maxjitterbuffer=1500
rtptimeout=60


As for the DTMF issue try to use rfc2833

in sip.conf define your extention

[XXXX]
username=XXXX
type=friend
secret=XXXXX
qualify=no
port=5060
nat=yes
mailbox=XXXX@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=device <XXXX>

Rich




 
Message: 1
Date: Wed, 19 Apr 2006 11:02:46 -0500
From: "Carlos Alberto Bernat Orozco" <cabo81@gmail.com>
Subject: [Asterisk-Users] Delayed voice for 10 secs
To: asterisk-users@lists.digium.com
Message-ID:
	<111c22550604190902o4b6e6ec1sa77ad0374ca69ec9@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi List !!

I have a lot a questions about this incredible tool but short is my time
to
learn it, so I apologize if my last question was too general. I got
another
more especific trouble. I administrating an ISP and I have my Asterisk
installed on a server for testing my network performance. I followed the
quick-start tutorial provided by voip-info.org (which I think it's very
useful) and configured two SJphones as extensions. My network is HFC
type
and the users surf the Internet by a Cable Modem (Motorola). When I
tried
this 2 softphones, the voice was delayed for 10 secs aprox. and the next
warning is jumping on my screen:

WARNING[26673]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported
on
codec gsm. Use RFC2833

I know I must change codecs in order to get the voice more fluency but I
don't know yet if I have to configure it on the Asterisk server (on
sip.conf)
or somewhere else (on the SJphones). Can you give me some info about it?
I
would appreciate a lot

Thanks

Carlos Bernat