asterisk users - May 2009 - asterisk 1.6.1.0 and dial plan changes

Hi all,

I have installed asterisk latest stable version 1.6.1.0, with dahdi 
driver (tdm410p). then i try to use my older 1.4 extensions.conf. . now 
it wont work with 1.6.

I managed to register my phone on asterisk. but i cant hear any dial 
tone on my phone.  these are my configs.  it will detect incoming calls 
and transfer the call to ext 312.  but sip phone users voice is not 
clear..., but sip phone user can hear the other party (PSTN) very clearly.

please help me to solve the issue. all work on asterisk 1.4.

[general]

port = 5060                    
bindaddr = 0.0.0.0          
context = sip       
disallow=all
allow=all
;allow=g729
;allow=gsm
allow=alaw
allow=ulaw
transfer=yes
tos=lowdelay
dtmfmode = rfc2833

[312]
type=friend                    ; Friends place calls and receive calls
context=sip2            ; Context for incoming calls from this user
secret=312
host=dynamic                ; This peer register with us
dtmfmode=rfc2833               ; Choices are inband, rfc2833, or info
username=312               ; Username to use in INVITE until peer registers
mailbox=312
qualify=yes
disallow=all
pickupgroup=1
allow=all
;allow=alaw                     ; dtmfmode=inband only works with ulaw 
or alaw!
;allow=gsm
;;canreinvite=no
;;progressinband=yes
;;reinvite=no
;;callerid=tharanga <312>


extensions.conf



channel.dadhi.conf

[channels]


signalling=fxs_ks
;toneduration=100
callwaiting=yes
threewaycalling=yes
callreturn=yes
echocancel=128,param1=32,param2=0,param3=14
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
busydetect=yes
busycount=2
hanguponpolarityswitch=yes
ringtimeout=8000
group=1
context=sip
immediate=yes
jitterbuffers=4
jbenable = yes
echocancel=yes
channel=>1-4
;overlapdial=yes
;pulsedial=yes
dtmfmode=rfc2833
;relaxdtmf=yes
;rxgain=10.0
;txgain=8.0


Many thanks
Tharanga

On Fri, May 29, 2009 at 1:22 AM, Tharanga <tharanga at roomsnet.com>
wrote:
> I managed to register my phone on asterisk. but i cant hear any dial
> tone on my phone. ?these are my configs. ?it will detect incoming calls
> and transfer the call to ext 312. ?but sip phone users voice is not
> clear..., but sip phone user can hear the other party (PSTN) very clearly.
You've mentioned like three different things, each of which you should
attack separately. I can give some tips on the SIP voice quality
issue:

* take a look at dsp.conf, and make a larger silencethreshold value. I
set mine to 1000.
* take a look at codecs.conf, and change vad => false

You don't say the kind of call you're making, but if you're using
MeetMe() I have more advice regarding voice quality with conference
rooms.

On Fri, May 29, 2009 at 1:22 AM, Tharanga <tharanga at roomsnet.com>
wrote:
> I managed to register my phone on asterisk. but i cant hear any dial
> tone on my phone.
What kind of phone? What kind of channel?

On Fri, May 29, 2009 at 1:22 AM, Tharanga <tharanga at roomsnet.com>
wrote:
> I have installed asterisk latest stable version 1.6.1.0, with dahdi
> driver (tdm410p). then i try to use my older 1.4 extensions.conf. . now
> it wont work with 1.6.
You pasted in sip.conf and dahdi config, but not your extensions.conf
What's not working with your extensions?

1. You should get a dial tone from SIP as soon as you pick up the phone or
press the call button.
2. show us output of "dahdi status" , "dahdi show channels"
and "sip show
peers" from your CLI.
This will give important clues.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, May 29, 2009 11:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 1.6.1.0 and dial plan changes

On Fri, May 29, 2009 at 1:22 AM, Tharanga <tharanga at roomsnet.com>
wrote:
> I managed to register my phone on asterisk. but i cant hear any dial
> tone on my phone.
What kind of phone? What kind of channel?

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