:
- Stagg Shelton [EMAIL PROTECTED] wrote:
I did ultimately force asterisk to the point where it will not accept
or
send rfc2833. I did this by modifying chan_sip.c in the function
Asterisk should not be sending an SDP with RFC-2833 in it when the dtmfmode=inband in sip.conf
rfc2833
Stagg Shelton
Sun, 02 Jul 2006 13:38:23 -0700
I answered my own question. My objective was reached with a simple
return statement on line 4384 of chan_sip.c in asterisk 1.2.9.1 ftp
download. The effect that this has is that asterisk will return a 200 OK
that indicates in the SDP that only
to make asterisk
do what I need it to. I'm hoping to have someone who is familiar with
chan_sip.c enlighten me as to whether or not this can be done, and what
functions I would need to modify in order to make it happen.
Thank You
Stagg Shelton
www.oneringnetworks.com
this and their NexTone Session Switch sends out dtmf
inband.
It sucks having to force asterisk to operate in this manner, but
hopefully asterisk implementation of rfc2833 will get the bugs worked
out, if they are in fact bugs, and not design desicions.
Stagg Shelton wrote:
I'm trying
If you want to try and get it working try reading through
http://lists.digium.com/pipermail/asterisk-users/2006-February/147198.html.
I had a tough time stamping out echo to far-end analog pstn connections.
Stagg Shelton
www.oneringnetworks.net
Imran Ahmed wrote:
Use:
modprobe wct4xxp
Also I using sip version 1.6.3.0067 and bootrom 3.1.2
Stagg Shelton
www.oneringnetworks.com
Anton Krall wrote:
What We are hearing here is that our local users say when they raise their
voice, sometimes they hear themselves back.. So I guess.. If We have RX and
TX setting on the polycoms
. If you are planning to build a
substantial network using wifi, the technical list of issues that await
you are too long to list.
All that negativity being said, I've had a blast doing my part to help
build out our network.
Best of luck to you.
Stagg Shelton
www.oneringnetworks.com
Juergen K
to the appropriate case. So far in FC4 I haven't had any
problems creating mixed case usernames as required for easy polycom
mass deployment.
Stagg Shelton
www.oneringnetworks.com
Noah I. Miller wrote:
Hi Matt -
I have the same problem. I'm running CentOS, which comes
with vsftpd, do you know
I forgot about one other issue we had with the 2850. The integrated
NICs caused interrupt issues with the TE411P. We had to disable the
integrated NICs, and installed dual port gigabit intel NIC.
Stagg Shelton
www.oneringnetworks.com
[EMAIL PROTECTED] wrote:
Alexander, Perhaps I'm wrong
that could be used to power the TDM400P.Aside from the power
issue, the platform seems rock solid.
Stagg Shelton
www.oneringnetworks.com
[EMAIL PROTECTED] wrote:
Hey everyone,
I've been doing a lot of research into a decent server for Asterisk
but I seem to be running and circles and now I
to a message previously with the exact trunk version that
I downloaded: SVN-trunk-r941
Stagg Shelton
www.oneringnetworks.com
Eric Bishop wrote:
Is this with the TE411P? Also what do you mean by "pulled
the zaptel trunk source"?
On 2/17/06, Stagg Shelton
[EMAIL PROTECTED] wrote:
point to point wireless. Voice is great though :)
Stagg Shelton
www.oneringnetworks.com
Adolfo R. Brandes wrote:
turby wrote:
yes, with last patch works well. thanks.
Glad to be of service!
Adolfo
___
--Bandwidth and Colocation provided
==
SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
28 channels configured.
Stagg Shelton
www.oneringnetworks.com
Andrew Kohlsmith wrote:
On Thursday 16 February 2006 11:11, Stagg Shelton wrote:
Here are my final configurations
zaptel trunk pulled 2/13/06
=pri_cpe
resetinterval=never
faxdetect=incoming
usecallerid=yes
echocancel=yes
echotraining=800
rxgain=4.5
txgain=-13.5
group=0
channel=1-23
Thank You for all of your pointers and support in this issue.
Stagg Shelton
www.oneringnetworks.com
resetinterval=never
faxdetect=incoming
usecallerid=yes
echocancel=yes
echotraining=800
rxgain=4.5
txgain=-13.5
group=0
channel=1-23
Thank You for all of your pointers and support in this issue.
Stagg Shelton
www.oneringnetworks.com
___
--Bandwidth
I am using asterisk 1.2.4 and zaptel 1.2.3. Also, I tried the latest
zaptel out of subversion.
Stagg Shelton
www.oneringnetworks.com
Isaac Xiao (KVB Kunlun Pty Limited) wrote:
What version
of Asterisk and Zaptel you were using? Did
you try latest Asterisk 1.2.4 and Zaptel
another data point to help me determine if the HW module is active.
More to come...
Stagg Shelton
www.oneringnetworks.com
Cory Andrews wrote:
Stagg - I know you get a full 128ms
tail of echo can on the Sangoma. I believe that on the TE411P, the
128ms tail is shared by all (4
kinds of
products that I could look into either Passive or Active.
Thanks
Stagg Shelton
www.oneringnetworks.com
Matt wrote:
try sangoma carrier grade 104d hardware EC card. we're using it ourself.
Best Regards
Matt
- Original Message -
From: "Anthony Rodgers" [EMAIL
?
--Stagg
www.oneringnetworks.com
Kevin P. Fleming wrote:
Stagg Shelton wrote:
I just implemented a system using a TE411P hardware echo cancellation
card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as
I always have. To my surprise calls out to the PSTN had a terrible
echo. 1
I just implemented a system using a TE411P hardware echo cancellation
card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as
I always have. To my surprise calls out to the PSTN had a terrible
echo. 1 - 2 second delay, and quite clear. The echo was so bad that I
had to remove
I'm hearing an echo when using a headset with my IP 501 / 601. The
phones are using BR 3.1.2 and SIP 1.6.3. I use tftp to configure the
phones. The sip.cfg is the default from polycom except for the
parameters required to connect the phones to asterisk.
I have absolutly no echo with the
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