[Asterisk-Users] softphone has problem to call out via X100P card

[Anton Krall]

I think Kris gave some very nice pointers... If you have any more questions
afterwars, please let me know.
 

-----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 10:24 p.m.
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] softphone has problem to call out via X100P
card

HI Anton, thanks for the quick reponse. For you information, I am new in
both * and Linux.

How do you include context in Zapata.conf for incoming calls? could you
please put the comment on my config?

How do I include the [outgoing] on my [from-sip] context? Should I just do a
cut whatever under [outgoing] and paste them under [from-sip]?

Or can I put two context under the same phone. exp:

[2000]
type=friend
username=2000
secret=2000abc
context=from-sip
context=outgoing <<<=== Is this allowed under * mailbox=100


>From: "Anton Krall" <akrall-lists at intruder.com.mx>
>Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
><asterisk-users at lists.digium.com>
>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
><asterisk-users at lists.digium.com>
>Subject: RE: [Asterisk-Users] softphone has problem to call out via 
>X100P card
>Date: Thu, 24 Feb 2005 22:05:30 -0600
>
>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)
>
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