Hello Jag:
With regards to DTMF, this may be a combination of one or both of the following:
1. The way the SIP phone sends DTMF
Resolution: Check how the DTMF is processed. That is:
a. SIP Info
b. Inband Audio
c. Both.
Play with that and see if it resolves the issue.
2. The way your Asterisk process the DTMF
a. Check what DTMF default mode is.
b. set it to rfc2833
c. or set it to inband (must have alaw or ulaw enabled)
d. play with the above and see how it works.
I too use Unlimitel. Stephan Monette & his guys are great! I have not
encountered any of these issues/problems after having properly configured the
way my sip phone handles dtmf. Most proprietary systems your asterisk
communicates with, require you to send as inband audio to their PBX.
With regards to:
I also noticed my Music on hold is not clear but rather choppy, didn't have
time to troubleshoot yet... but maybe the 2 are releated?
Well... I had exactly the same issue. The way I resolved this is by having a
single line FXO (clone x100p or x101p) installed, which behaves like a timer.
MoH is a feature that heavily depends on a timing device to my understanding.
The software based timing device (ztdummy) is probably what you are using. I
had configured my system to ztdummy when I originally encountered this issue.
I've noticed that with Kernel 2.6 and faster processors with USB2.0, the
ztdummy works well.
The instant when I used an FXO with a more or less better timing device than
the software based ztdummy timing device, the choppy music issue was resolved.
If you want to know more about what this timing device is... read one of the
longest threads in this group on "Timing Devices" :). I've learnt a LOT from
everyone who pitched in their 2cents with regards to timing device.
Other folks might have similar experiences - but I can only speak for myself,
which I believe, may also solve your MoH issue.
Cheers!
Reza.
----- Original Message -----
From: Jag Basrai
To: [email protected]
Sent: Wednesday, July 19, 2006 11:22 PM
Subject: [on-asterisk] problems with sending signals.
Hi Guys,
Today I was calling a company that has an auto attendant, and I was calling
via my asterisk box using unlimitel, and I found that I have to wait for the
person to stop talking before I can press the extension I wish to use. If I
dial the extension beforehand it either does not register on their end, or it
says incorrect extension was entered. Any idea why my asterisk box is not
sending the correct signal while the other attendant is still speaking? I am
using voip (unlimitel) with aastra 480i's??? I think it is some kind of 2 way
audio problem..
I also noticed my Music on hold is not clear but rather choppy, didn't have
time to troubleshoot yet... but maybe the 2 are releated?
Also would like to thank everyone involved in setting up todays workshop on
OpenWRT, great job guys... I learned alot :)
-- Jag
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