Ho Chan
2005-Feb-24 20:26 UTC
[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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Anton Krall
2005-Feb-24 21:05 UTC
[Asterisk-Users] softphone has problem to call out via X100P card
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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ho Chan
Sent: Jueves, 24 de Febrero de 2005 09:26 p.m.
To: asterisk-users@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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Kris Stark
2005-Feb-24 21:20 UTC
[Asterisk-Users] softphone has problem to call out via X100P card
Ho Chan wrote:> 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)You need to allow the softphones to go to the context that allows dialling out. See below - inline comments...> 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)You need to define a context for the zap inbound channel to go to - see below...> Zapata.conf > > language=en[snip]> channel=1context=from-sip The above is not quite recommended, but would work with the config that you have... Ideally, you'd have an incoming context in your extensions.conf, and the context in zaptel would then point into that context.> ------------------------------------------------------------------------------------------------- > > Sip.conf > > [general] > port = 5060 > bindaddr = 0.0.0.0 > allow=all > > context = outgoing > > [2000] > type=friend > [snip] > context=from-sip > mailbox=100This sends all your calls from the softphone into the "from-sip" context in extensions.conf.> Extension.conf > > [general] > static=yes > writeprotect=yes > > [outgoing] > ignorepat => 9 > exten => _9NX.,1,Dial(ZAP/1/${EXTEN:1},60,t) > exten => _9NX.,2,CongestionNone of your calls ever get into this context... You would want to have a "include => outgoing" in your from-sip context to be able to dial out. However, with the way your extensions.conf is set up right now that would be a security risk since to be able to call in you would have to have your inbound calls from the PSTN going into this same context...> [from-sip] > exten => 2000,1,NoOp("call for "${EXTEN}) > exten => 2000,2,Dial(SIP/2000,20,tr) > exten => 2000,3,Voicemail(u2000) > exten => 2000,102,Voicemail(b2000) > exten => 2000,103,Hangup[snip]> exten => 2999,1,VoicemailMain(${CALLERIDNUM}) > > ; Call straight to extension 2001 > > exten => s,1,Answer > exten => s,2,Dial(SIP/2001,20,tr) > exten => s,3,Voicemail(u2001) > exten => s,4,Voicemail(b2001)Basically, try something like: [inbound] ; Since you want all your calls from the PSTN to go to ext 2001 exten => s,1,Answer exten => s,2,Dial(SIP/2001,20) exten => s,3,Voicemail(u2001) exten => s,103,Voicemail(b2001) [outbound] ignorepat => 9 exten => _9NX.,1,Dial(ZAP/1/${EXTEN:1},60,t) exten => _9NX.,2,Congestion [from-sip] include => outbound 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}) Kris -------------- next part -------------- A non-text attachment was scrubbed... Name: Kris.Stark.vcf Type: text/x-vcard Size: 306 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050224/6300b4ea/Kris.Stark.vcf