[Asterisk-Users] softphone has problem to call out via X100P card
Anton Krall
akrall-lists at intruder.com.mx
Thu Feb 24 21:05:30 MST 2005
I think it's a context problem.
I didn't see any context on zapata.conf so your incoming callsmight be going
nowever, check that zapata.conf includes a context to your main main.
Also, your phones have the context from-sip but your dialout is on context
outgoing, so your phones have no way of knowing how to dialout, include your
outgoing on your from-sip context and try again.
If you need more help, please let me know.
Anton Krall
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ho Chan
Sent: Jueves, 24 de Febrero de 2005 09:26 p.m.
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] softphone has problem to call out via X100P card
Hi all,
I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC.
With the following configuration, I can use one softphone (2000) to call the
other one (2001) and/or the voicemail at 2999.
Here is my problem:
1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via
X100P card, I got busy tone. (i.e. I want to use the phone line which is
connected to the X100P to call out)
2. When I use my cell phone to call the phone line which is connected
to X100P, it just rings for 4 times then hang up on me. (i.e. Asterisk never
answer the phone)
Anybody can verify my configuration? I am very new to *.
Thanks
Terry
----------------------------------------------------------------------------
---------------------
Zapata.conf
language=en
busydetect=yes
busycount=4
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
usecallerid=yes
hidecallerid=no
echocancel=yes
echocancelwhenbridged=yes
usecallingpres=yes
canpark=yes
cancallforward=yes
callreturn=yes
rxgain=0.0
txgain=0.0
immediate=no
signalling=fxs_ks
callerid=asreceived
channel=1
----------------------------------------------------------------------------
---------------------
Sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
allow=all
context = outgoing
[2000]
type=friend
username=2000
secret=2000abc
auth=md5
nat=yes
host=dynamic
reinvite=no
canreninvite=no
qualify=1000
callerid="Terry Chen" <2000>
disallow=all
allow=gsm
context=from-sip
mailbox=100
[2001]
type=friend
username=2001
secret=2001abc
auth=md5
nat=yes
host=dynamic
reinvite=no
canreninvite=no
qualify=1000
callerid="xx xxx" <2001>
disallow=all
allow=gsm
context=from-sip
mailbox=101
----------------------------------------------------------------------------
---------------------
Extension.conf
[general]
static=yes
writeprotect=yes
[outgoing]
ignorepat => 9
exten => _9NX.,1,Dial(ZAP/1/${EXTEN:1},60,t)
exten => _9NX.,2,Congestion
[from-sip]
exten => 2000,1,NoOp("call for "${EXTEN})
exten => 2000,2,Dial(SIP/2000,20,tr)
exten => 2000,3,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup
exten => 2001,1,NoOp("call for "${EXTEN})
exten => 2001,2,Dial(SIP/2000,20,tr)
exten => 2001,3,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
; Call straight to extension 2001
exten => s,1,Answer
exten => s,2,Dial(SIP/2001,20,tr)
exten => s,3,Voicemail(u2001)
exten => s,4,Voicemail(b2001)
_________________________________________________________________
Get 10Mb extra storage for MSN Hotmail. Subscribe Now!
http://join.msn.com/?pgmarket=en-hk
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list