Tejas Shah
2005-Nov-30 22:34 UTC
[Asterisk-Users] two sip phone communication using asterisk server
hi,
I am a newbie to asterisk. I installed a asterisk server to make
communication between 2 X-Lite's SIP based phones. I made following
configuration in sip.conf :
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[2000]
type=friend ; This device takes and makes calls
username=2000 ; Username on device
secret=9overthruster7 ; Password for device
host=dynamic ; This host is not on the same IP addr every time
context=from-sip ; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[2001] ; Duplicate of 2000, except with different auth data
type=friend
username=2001
secret=11bbanzai9
host=dynamic
context=from-sip
mailbox=101
and following configuration in extension.conf :
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[bogon-calls]
;
; Take unknown callers that may have found
; our system, and send them to a re-order tone.
; The string "_." matches any dialed sequence, so all
; calls will result in the Congestion tone application
; being called. They'll get bored and hang up eventually.
;
exten => _.,1,Congestion
[from-sip]
;
; If the number dialed by the calling party was "2000", then
; Dial the user "2000" via the SIP channel driver. Let the number
; ring for 20 seconds, and if no answer, proceed to priority 2.
; If the number gives a "busy" result, then jump to priority 102
;
exten => 2000,1,Dial(SIP/2000,20)
;
; Priority 2 send the caller to voicemail, and gives the "u"navailable
; message f or user 2000, as recorded previously. The only way out
; of voicemail in this instance is to hang up, so we have reached
; the end of our priority list.
;
exten => 2000,2,Voicemail(u2000)
;
; If the Dialed number in priority 1 above results in
; a "busy" code, then Dial will jump to 101 + (current priority)
; which in our case will be 101+1=102. This +101 jump is built
; into Asterisk and does not need to be defined.
;
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup
;
; Now, what if the number dialed was "2001"?
;
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
;
; Define a way so that users can dial a number to reach
; voicemail. Call the VoicemailMain application with the
; number of the caller already passed as a variable, so
; all the user needs to do is type in the password.
;
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
now my problem is when i m starting asterisk server both Sip phones are
showing registration. when i make call from any of PC following error occurs on
the screen of asterisk server :
pbx.c:1731: can not find extension context 'from-sip'
when i close asterisk server communication is taking place beween both phones.
now i m stuck with this error. can anybody give me guidance on how to solve
this problem?
thanks
tejas
---------------------------------
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Alejandro Vargas
2005-Dec-01 05:08 UTC
[Asterisk-Users] two sip phone communication using asterisk server
2005/12/1, Tejas Shah <tejas705@yahoo.com>:> I am a newbie to asterisk. I installed a asterisk server to make > communication between 2 X-Lite's SIP based phones. I made following > configuration in sip.conf :For newbies (like me) a good start is to use amp or install directly asteriskathome. It solves all the problems of configuring and creating extensions. Then you can start lerning how to do the difficult things. -- Alejandro Vargas