[Asterisk-Users] Ztdummy - is it requirement?

[Marko Rakar]

My asterisk is located on 100mbit switched network (HP procurve 2524
switches); on asterisk server itself there is no sound hardware. I have
installed single hisax ISDN BRI adapter which is connected to my PBX.

As for clients, I use mediatrix FXS units with SIP protocol.

One of the mediatrix units is located on local LAN, second one on our
second location which is connected with 2mbit link. Same thing happens
with both devices.

Asterisk behaves the same in all circumstances; wheter I call from
mediatrix to ISDN or from ISDN to mediatrix (does not matter which
device) sound coming from mediatrix will be OK but sound going to
mediatrix is either choppy, noisy or completely cut-off. I have made all
adjustments on mediatrix which seemed possible (including turning
silence supression off, turning VAD off and playing with a lot of other
stuff). I do no know how (if at all possible) is to turn silence
supression on asterisk/hisax device. I have checked interrupts, changed
card position, reinstalled linux/asterisk.

Interesting is that when I work with echo test from either mediatrix
unit or from isdn it works perfectly. If I talk from mediatrix to
mediatrix it is also OK, If I route call from one ISDN B channel to
another it is also OK. 

So I was thinking that maybe there is problem with RTP stream to
mediatrix from my ISDN card and this might be related to the fact that
there is no zaptel hardware which gives clock information to RTP. (I
have installed ztdummy in the meantime but since I am alone in the
office it is a little hard to make a call :)


----
Sometimes you're the bug, sometimes you're the windshield.

mailto:marko at printel.hr
http://printel.hr  

-----Original Message-----
From: Philipp von Klitzing
[mailto:klitzing at pool.informatik.rwth-aachen.de] 
Sent: Saturday, April 03, 2004 8:54 PM
To: asterisk-users at lists.digium.com
Cc: Marko Rakar
Subject: Re: [Asterisk-Users] Ztdummy - is it requirement?


Hi!

> I am interested to learn if I need to have ztdummy installed if I do 
> not have any zaptel hardware in my machine?

No, not necessarily. You'll only need it if you want to use MeetMe 
conferencing. Look here:

http://www.voip-info.org/tiki-index.php?page=Asterisk+timer

> I have found a lot of references with RTP problems which were related 
> to RTP timing (or lack of it).
> 
> My problem is that voice coming from SIP hardware is OK, but voice 
> going from asterisk to SIP hardware is choppy, full of noise or 
> completely cut-off. Am I going to solve my problem with ztdummy (which

> btw. I can not compile but I see that this is also common problem)?

Most likely you have a slience suppression issue here, and that is not 
related to ztdummy. If you use X-Lite as SIP client then you'll need to

make sure you have "Transmit Silence" set to YES. For other devices like

Grandstream phones etc you'll find different names being used for the 
same thing (for example "VAD"). You can test this by constantly creating

sound on your side when hearing "choppy sound" - if that fixes the 
problem then you have a silence suppression issue.

As for the noise I can only guess: 

- make sure you use a good SIP client (and a GOOD soundcard if this is a

softclient; it might help if you told the list WHICH client you are 
working with, by the way, that'll cut down the guesswork)
- don't ever run X-Windows on your Asterisk server
- check your network and tell us what kind of connection you have
between 
phone and Asterisk server; maybe your upload link Ast --> phone is too 
thin & too packed?
- are there rules as to when you get a) choppy sound, b) no sound and c)

noise? Do you see that problem also with 1. voicemail and 2. music-on-
hold?
- what channel (technology) are you calling with your SIP device?

> What changes I have to make in modules.conf file in order to start 
> using ztdummy?

Usually none - but see URL above for details: You'll need to edit the 
zaptel Makefile.

Cheers, Philipp