asterisk users - Jan 2005 - CallerID for incoming SIP calls to Asterisk connected phone

I'm testing a bunch of stuff before we implement our system. 

I've got a SIP account and Grandstream phones.  We haven't started using
asterisk yet.  Generally we've got good voice quality from all the
offices except:

a) We get a lot of echo in the first 10 seconds or so of the call, only
on the VOIP calling end.  The callee says the speech sounds normal.  To
the caller, the first Hello is almost intelligible with echo. 

b) The first part of an abrupt statement from one party gets
"clipped".
In conversation, when talking switches from one party to the other, a
tiny bit of speach gets clipped.

c) If both parties talk at once there is a bit of dropout. 

We'd like to improve the voice quality in these respects.  Otherwise the
voice quality is excellent.  I've been told it is better than the
traditional system several times.  

Questions:

a) Are certain codecs better than others at quickly getting the echo
cancellation setup ?  Is there a way to get the echo out of the call
immediately ?  (Is there a document explaining the features and pitfalls
of all the codecs somewhere ?)  

b) Is there a way to eliminate the speech clipping when speakers change
or both talk at once ?  I've read about asterisk injecting noise and/or
sending packets in the absence of speech.  Would that help ?  Is this
what the Grandstream "Silence Suppression" is about ?
   
c) How does one know where to set the following:
	
iLBC frame size: 20ms 30ms 
iLBC payload type: (between 96 and 127, default is 98)
Silence Suppression: No Yes 
Voice Frames per TX: (up to 10/20/32/64 for G711/G726/G723/other codecs
respectively) 
Layer 3 QoS: (Diff-Serv or Precedence value)
Layer 2 QoS: 802.1Q/VLAN Tag      802.1p priority value  (0-7) 

d) One place we've really got a problem is when we use a Grandstream in
a big echoy (sp!) room.  We seem to get echo from the room into the call
which seems to fool the echo cancellation.  Any ideas on how to get
around this problem ?

d) How is asterisk going to change our sound quality  when it is added
between the phones and the SIP provider ?  Does it have features that
will help with the echo and clipping and if so, how much improvement
should we expect ?

Thanks. 


-- 
Kim Lux,  Diesel Research Inc.

Hello all,

I'm having a problem with getting incoming callerid to a lan-connected 
phone.

The Asterisk server is connected to the Internet, and a Grandstream 
BT101 phone on a lan interface:
INTERNET ----(eth0) Asterisk (eth1) ---- Grandstream (192.168.1.51)
The phone registers with the Asterisk server as ext 20.

I can initiate and receive calls from the Grandstream phone fine.

The Asterisk server has a sipphone.com registered account.  When a SIP 
call comes in from outside, the call completes fine, but the phone 
always shows the telephone number of my Asterisk server, not the calling 
party's SIP number.  What's wrong?

What I really want is that for inbound calls, I see the callerid of the 
SIP phone initiating the call.

Here are the (hopefully) relevant parts in the config files...

In sip.conf:
-----------
register => 1747xxxxxxx:mypassword@proxy01.sipphone.com/1747xxxxxxx

[sipphone]
context=from-sip-external
type=friend
secret=sip_password
username=1747xxxxxxx
;host=proxy01.sipphone.com
host=198.65.166.131
callerid="My Name <1747xxxxxxx>:
qualify=no
reinvite=no
canreinvite=no
insecure=very

[20]
context=from-sip-internal
type=friend
callerid=20
username=20
mailbox=20
secret=xxxx
host=dynamic
defaultip=192.168.1.51
canreinvite=no
dtmf=info
dtmfmode=rfc2833
;
disallow=all
allow=ulaw
allow=alaw
allow=ilbc


In extensions.conf:
------------------

[globals]
TRUNK=Zap/1  ; FXO interface
SIPPHONEUSERID=1747xxxxxxx

[from-sip-external]
exten => ${SIPPHONEUSERID},1,SetCIDName(SIP - ${CALLERIDNAME})
exten => ${SIPPHONEUSERID},2,Dial(SIP/20,15)
exten => ${SIPPHONEUSERID},3,Goto(mainmenu,s,1)
exten => ${SIPPHONEUSERID},4,Hangup
...

Any suggestions/help would be greatly appreciated.

Thanks,
Alvin

PS: Please cc me directly on replies:
   a a n (at) crlogic (dot) com