Hi ,
I am receiving bad call quality issues when it comes to outgoing calls. We are using a digium BRI-410P with 4 lines (8 channels) . Is there anyway to tweak up the quality ?
Location : Italy ( Fastweb)
Client phones : GXP 2000
Asterisk version: 1.4.18.1
This is my sip.conf
[general]
videosupport=yes
pedantic=yes
context=default ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
tos_sip=cs3 ; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
maxexpiry=120 ; Maximum allowed time of incoming registrations
; and subscriptions (seconds)
disallow=all ; First disallow all codecs
allow=alaw ; Allow codecs in order of preference
allow=ulaw
allow=ilbc
allow=h261
allow=h263
allow=h263p
useragent=Asterisk PBX ; Allows you to change the user agent string
matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.
rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
; on the audio channel
; when we're on hold (must be > rtptimeout)
rtpkeepalive=20 ; Send keepalives in the RTP stream to keep NAT open
; (default is off - zero)
;--------------------------- SIP DEBUGGING ---------------------------------------------------
sipdebug = no ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
nat=no ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason wants Asterisk to
; stay in the audio path, you may want to turn this off.
; Insert server specific include files below
#include "server_specific_sip.conf" ; Specific file for each country
And this is my misdn.conf
[general]
misdn_init=/etc/misdn-init.conf
debug=1
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
ntkeepcalls=no
tracefile=/var/log/asterisk/misdn.log
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh
[default]
context=misdn
language=en
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
pmp_l1_check=no
reject_cause=16
need_more_infos=no
nttimeout=no
method=standard
overlapdial=yes=5
dialplan=2
localdialplan=0
cpndialplan=2
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
presentation=-1
screen=-1
echocancel=yes
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
max_incoming=-1
max_outgoing=-1
[intern]
ports=1,2,3,4
context=misdn
msns=*
And my misdn-init.conf
card=1,0x4
te_ptp=1,2,3,4
poll=128
dsp_poll=128
dsp_options=0
dtmfthreshold=100
debug=0
Thanks !