Hi all! I'm new to asterisk and I'm trying a simple config with: - Debian GNU/Linux (unstable) - last version of Asterisk - a X100P card I have a problem with dial out from a SIP software phone (XLITE) to a public number (ex. my mobile phone), asterisk start the call, but nothing happen... If I run "ztmonitor 1" I can see the right RX level and if I try to make a call with an analog standard phone connected to the second plug of the X100P, I can see the RX level going UP and down normally, and I can also hear my voice during a call. Otherwise, when I try to dial out from XLITE, when I start the call the RX level go to 0 and I can only hear the numbers of che called number but I can hear nothing on RX and the line is "locked" until I remove the wcfxo kernel module; in Italy we must wait for a tone before starting the call.... Is there anyone here with an idea for my problem ? Thanks in advance. Piero Baudino
On Wed, Aug 17, 2005 at 02:20:17PM +0200, Piero Baudino wrote:> Hi all! > > I'm new to asterisk and I'm trying a simple config with: > - Debian GNU/Linux (unstable) > - last version of Asterisk > - a X100P card > > I have a problem with dial out from a SIP software phone (XLITE) to a > public number (ex. my mobile phone), asterisk start the call, but nothing > happen...CLI trace?> If I run "ztmonitor 1" I can see the right RX level and if I try to make a > call with an analog standard phone connected to the second plug of the > X100P, I can see the RX level going UP and down normally, and I can also > hear my voice during a call. > Otherwise, when I try to dial out from XLITE, when I start the call the > RX level go to 0 and I can only hear the numbers of che called number but > I can hear nothing on RX and the line is "locked" until I remove the > wcfxo kernel module; in Italy we must wait for a tone before starting the > call.... > > Is there anyone here with an idea for my problem ?asterisk -rvvv zap show channels and report what you see, and what happens when you try to call. Have you configured an asterisk zap channel for the phone? Are you calling through it? -- Tzafrir Cohen | tzafrir@jbr.cohens.org.il | VIM is http://tzafrir.org.il | | a Mutt's tzafrir@cohens.org.il | | best ICQ# 16849755 | | friend
Hi Tzafrir,
thanks for your reply...
Here is what happens when I make the call:
pbx*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo incoming it
1 incoming it
-- Executing NoOp("SIP/6601-5d39", ""Call for
"91234567") in new
stack
-- Executing Dial("SIP/6601-5d39", "Zap/1/91234567") in
new stack
-- Called 1/91234567
-- Zap/1-1 answered SIP/6601-5d39
-- Hungup 'Zap/1-1'
== Spawn extension (x-lite, 912334567, 2) exited non-zero on
'SIP/6601-5d39'
-- Unregistered SIP '6601'
pbx*CLI> exit
The Hangup happens when I hangup from XLITE.
Here is my conf:
/etc/asterisk/zapata.conf
[channels]
language=it
signalling=fxs_ks
context=incoming
channel=>1
/etc/asterisk/extensions.conf
[incoming]
exten => s,1,Dial(SIP/6601&SIP/6602&SIP/6603,20,tr) ; corresponding
clients must be configured in sip.conf
exten => s,2,Playback(vm-goodbye)
exten => s,3,Hangup
[outgoing]
exten => _9X.,1,NoOp("Call for "${EXTEN})
exten => _9X.,2,Dial(Zap/1/${EXTEN})
[x-lite] ; Note: SIP extensions are defined here as "66" followed by
any
two digits
exten => _66XX,1,NoOp("Call for "${EXTEN})
exten => _66XX,2,Dial(SIP/${EXTEN})
exten => _66XX,3,Congestion
include => outgoing
/etc/asterisk/sip.conf
port=5060
context=default
srvlookup=yes
dtmfmode=inband
allow=aLaw
allow=uLaw
allow=gsm
[6601]
type=friend
secret=password
host=dynamic
;dtmfmode=rfc2833
context=x-lite
callerid="Piero" <6601>
allow=aLaw
allow=uLaw
allow=gsm
Thanks.
PieroB