Try the following configs:
/etc/asterisk/zaptel.conf
fxsks=1 loadzone=uk defaultzone=uk
/etc/asterisk/zapata.conf
languages=en context=inbound-analog signalling=fxs_ks
; I always create dial groups for making outbound calls, you can use the specific channels as well
group = 1
channel => 1
/etc/asterisk/sip.conf
[general] ; I generally prefer gsm and ulaw, you can allow any codecs you like disallow=all allow=gsm allow=ulaw
; use your IP address in the bind address or leave as 0.0.0.0 to bind to all active interfaces
port=5060
bindaddr=0.0.0.0
; set your tos - see www.voip-info.org command reference for tos values tos =0x10
;next create an entry for your SIP phones
; you can specify username and secret or you can set a very explicit permit.
; canreinvite, no=asterisk remains in media path, yes=asterisk CAN step out of media path
; if you have problems with authentication, try removing the username, secret, and permit lines
; and setting host=a.b.c.d where a.b.c.d is the ip address of the SIP client
; the example permit will permit any clients with 10.0.0.0 255.255.255.0 address space
[2001] type=friend username=2001 secret=2001 host=dynamic permit=10.0.0.0/8 canreinvite=no context=intern callerid=Test Caller mailbox=2001 nat=yes
/etc/asterisk/extensions.conf
[general]
static=yes writeprotect=yes
[globals]
; used for global variables, which in this basic example, we'll completely ignore
[outbound-analog]
exten => _X.,1,Dial(Zap/g1/${EXTEN},60)
exten => _X.,2,Hangup[inbound-analog]
exten => s,1,Dial(SIP/2001,20) exten => s,2,Voicemail(u2001) exten => s,3,Hangup exten => s,102,Voicemail(b2001) exten => s,103,Hangup
[local]
; Note we don't send local callers to Voicemail in this example
exten => 2001,1,Dial(SIP/2001) exten => 2001,2,Hangup exten => 2001,102,Hangup
exten => 2999,1,Answer exten => 2999,2,Wait(1) exten => 2999,3,VoiceMailMain exten => 2999,4,Hangup
[intern]
include => local include => outbound-analog
/etc/asterisk/voicemail.conf
[EMAIL PROTECTED] attach=yes maxmessage=300 maxgreet=60
[default] 2001 => 1234,John Doe,[EMAIL PROTECTED]
This should give you a very basic system with a SIP phone client, one outside line via X100P, and voicemail. the Sip client will be able to call voicemail using 2999, and any other sip clients you configure by dialing their extension. When someone calls the analog number from the outside world, the sip client at 2001 will ring, if no one answers, the caller will be sent to leave a voicemail message, if 2001 is busy, the caller will be sent to voicemail with a prompt indicating the caller is busy.
Hope this helps.
-Chris
At 12:08 PM 3/7/2004, you wrote:
Thanks for your help David
Your configs are a little to complicated for this complete asterisk newbie though.
All i am actually after is how to get a sip phone to ring when the X100P is dialed on out landline, and how to get a sipphone to dial out through the X100P.
I have saved all your configs and had a trawl through them though.
I am a great believer in start simple then build it up and step by step it seems simple in the end but I keep stumbling on this task. once i have this i will look at call parking,conferencing (all the fun stuff) etc.. but at the moment all i would like to acheive is bridging the gap from sip to BT :-) IF you have any quick pointers to help me acheive that I would be very pleased.
Thanks again for taking the time to reply (especially on a sunday evening with the roast going cold)
Simon
David J Carter wrote:
Simon,
Caller ID does not work in the UK, well not on my BT or Telewest line's.
Have a look at my sample configs http://www.codepipe.com/id25.htm , I am also in the UK and these work for me.
Give me a call if ya want to chat about it.
Regards
Dave
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Simon Chappell Sent: 07 March 2004 16:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X100P dial in/out to sip phones
Hello all
I have recently stumbled accross voip and asterisk. We have a small network of vpns running in the uk. I have managed to get the sip phones dialing each other through asterisk and it is working great. (we are having long free conversations and that is something to get excited about).. My problem is that I cannot get the X100P i recently bought to dial out or do anything with incoming calls. I did loads of googling and found this snippet that made the zaptel card moan at me about callerid ask me to type a number then do nothing but offer silence.. [inbound-analog] exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,2,Macro(record-on,${PHONE1},${CALLERIDNUM}) exten => s,3,PrivacyManager exten => s,4,Dial(${PHONE1},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten => s,7,Playback(new/hello) exten => s,8,Playback(new/marisa-john-not-in-momnt) exten => s,9,Playback(new/theyre-rattlesnake-wrstling) exten => s,10,Voicemail(u${PHONE1VM}) exten => s,11,Hangup exten => s,108,Wait(2) exten => s,109,Voicemail(b${PHONE1VM}) exten => s,110,Hangup If i rem out that and run asterisk with -vvg i get this when i dial in to the x100p Mar 7 16:43:41 NOTICE[245776]: chan_zap.c:4624 ss_thread: Got event 2 (Ring/Answered)... Mar 7 16:43:44 NOTICE[245776]: chan_zap.c:4624 ss_thread: Got event 2 (Ring/Answered)... Mar 7 16:43:44 NOTICE[245776]: chan_zap.c:4624 ss_thread: Got event 2 (Ring/Answered)... Mar 7 16:43:49 WARNING[245776]: chan_zap.c:4695 ss_thread: CallerID returned with error on channel 'Zap/1-1' Mar 7 16:43:49 WARNING[245776]: pbx.c:1778 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler
So i feel i am getting there.. I would like the extensions to dial out and ring when the line rings.. can anyone give me a clue or point me in the right direction
I am in the UK by the way if that makes a difference.
Many thanks in advance
Simon
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-- Kind Regards
Simon Chappell url : www.isnsuk.com email : [EMAIL PROTECTED] PH: 01403 268474
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