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...
[snip]Zapata.conf
language=en
channel=1
context=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=100
This 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/${EXTEN:1},60,t) exten => _9NX.,2,Congestion
None of your calls ever get into this context... You would want to have a "include => outgoing" in your from-sip context to be able to dial out. However, with the way your extensions.conf is set up right now that would be a security risk since to be able to call in you would have to have your inbound calls from the PSTN going into this same context...
[from-sip]
exten => 2000,1,NoOp("call for "${EXTEN})
exten => 2000,2,Dial(SIP/2000,20,tr)
exten => 2000,3,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup
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 "${EXTEN})
exten => 2000,2,Dial(SIP/2000,20,tr)
exten => 2000,3,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangupexten => 2001,1,NoOp("call for "${EXTEN})
exten => 2001,2,Dial(SIP/2000,20,tr)
exten => 2001,3,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangupexten => 2999,1,VoicemailMain(${CALLERIDNUM})Kris
begin:vcard fn:Kris Stark n:Stark;Kris org:Dataflow adr:Suite B;;401 E State St;Ithaca;NY;14850;USA email;internet:[EMAIL PROTECTED] title:IT Manager tel;work:+1 607 272 8589 tel;fax:+1 607 272 8634 tel;cell:+1 607 768 4401 x-mozilla-html:FALSE url:http://www.goDataflow.com version:2.1 end:vcard
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