Asterisk T1 outgoing call not ringing


#1

HI i setup a T1 connection, every think work fine, but user can hear the remote party ring when they make an outbound call.

this is my system conf

Autogenerated by /usr/sbin/dahdi_genconf on Thu Jul 7 13:51:53 2011

If you edit this file and execute /usr/sbin/dahdi_genconf again,

your manual changes will be LOST.

Dahdi Configuration File

This file is parsed by the Dahdi Configurator, dahdi_cfg

Span 1: WCT1/0 “Digium Wildcard TE110P T1/E1 Card 0” (MASTER)

termtype: te

span=1,1,0,d4,ami
e&m=1-24
echocanceller=mg2,1-24

Global data

loadzone = us
defaultzone = us


and this is my chan_dahdi.conf

; Autogenerated by /usr/sbin/dahdi_genconf on Thu Jul 7 13:51:53 2011
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
;

; Span 1: WCT1/0 “Digium Wildcard TE110P T1/E1 Card 0” (MASTER)
[channels]

context=from-pstn
signalling=em_w
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
group=0
channel => 1-24
;Uncomment these lines if you have problems with the disconection of your analog lines
;busydetect=yes
;busycount=3
immediate=no
#include dahdi-channels.conf
#include chan_dahdi_additional.conf


#2

Try calling Progress() before Dial().

NB. This may start charging if the incoming side is external, which I believe is why Asterisk was changed to make this call required before early media would be passed through.


#3

Thanks David, i will try but believed that this problem is something in the dahdi conf, one wait that i solved this problem is adding the option r in the dial()


#4

You have channel associated signalling, which limits the number of call states that can be signalled. I am assuming that it can only signal ringback in band. If that is the case, you are not going to get 180 Ringing on the SIP(?) side as the result of anything that happens on the circuit switched side.

By using r on Dial, you are forcing the sending of AST_CONTROL_RINGING, and therefore 180 Ringing, even though no out of band ringing indication has arrived. This will cause the actual SIP phone to generate ring back tone.

What I’m suggesting is that using Progress will force the use of early media on the caller’s side, which will, hopefully allow it to hear the actual ring back tone arriving from the public network.


#5

OK thanks david for ur clear answer