<DIV id=RTEContent> <DIV id=RTEContent> <DIV id=RTEContent>hi,</DIV> <DIV> </DIV> <DIV> 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 :</DIV> <DIV> </DIV> <DIV><FONT color=#003366>[general]<BR><BR>port = 5060 ; Port to bind to (SIP is 5060)<BR>bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)<BR>allow=all ; Allow all codecs<BR>context = bogon-calls ; Send SIP callers that we don't know about here<BR><BR>[2000]<BR><BR>type=friend ; This device takes and makes calls<BR>username=2000 ; Username on device<BR>secret=9overthruster7 ;
Password for device<BR>host=dynamic ; This host is not on the same IP addr every time<BR>context=from-sip ; Inbound calls from this host go here<BR>mailbox=100 ; Activate the message waiting light if this<BR> ; voicemailbox has messages in it<BR><BR>[2001] ; Duplicate of 2000, except with different auth data<BR><BR>type=friend<BR>username=2001<BR>secret=11bbanzai9<BR>host=dynamic<BR>context=from-sip<BR>mailbox=101</FONT></DIV> <DIV><FONT color=#003366></FONT> </DIV> <DIV><STRONG>and following configuration in extension.conf :</STRONG></DIV> <DIV><STRONG></STRONG> </DIV> <DIV><FONT
color=#003366>[general]<BR><BR>static=yes ; These two lines prevent the command-line interface<BR>writeprotect=yes ; from overwriting the config file. Leave them here.<BR><BR>[bogon-calls]<BR><BR>;<BR>; Take unknown callers that may have found<BR>; our system, and send them to a re-order tone.<BR>; The string "_." matches any dialed sequence, so all<BR>; calls will result in the Congestion tone application<BR>; being called. They'll get bored and hang up eventually.<BR>;<BR><BR>exten => _.,1,Congestion <BR><BR>[from-sip]<BR><BR>;<BR>; If the number dialed by the calling party was "2000", then<BR>; Dial the user "2000" via the SIP channel driver. Let the number<BR>; ring for 20 seconds, and if no answer, proceed to priority 2.<BR>; If the number gives a "busy" result, then jump to priority 102<BR>;<BR><BR>exten => 2000,1,Dial(SIP/2000,20)<BR><BR>;<BR>; Priority 2 send the caller to voicemail, and gives the "u"navailable<BR>; message f or user
2000, as recorded previously. The only way out<BR>; of voicemail in this instance is to hang up, so we have reached<BR>; the end of our priority list.<BR>;<BR><BR>exten => 2000,2,Voicemail(u2000)<BR><BR>;<BR>; If the Dialed number in priority 1 above results in<BR>; a "busy" code, then Dial will jump to 101 + (current priority)<BR>; which in our case will be 101+1=102. This +101 jump is built<BR>; into Asterisk and does not need to be defined.<BR>;<BR><BR>exten => 2000,102,Voicemail(b2000)<BR>exten => 2000,103,Hangup<BR><BR>;<BR>; Now, what if the number dialed was "2001"?<BR>;<BR><BR>exten => 2001,1,Dial(SIP/2001,20)<BR>exten => 2001,2,Voicemail(u2001)<BR>exten => 2001,102,Voicemail(b2001)<BR>exten => 2001,103,Hangup<BR><BR>;<BR>; Define a way so that users can dial a number to reach<BR>; voicemail. Call the VoicemailMain application with the<BR>; number of the caller already passed as a variable, so<BR>; all the user needs to do is type in the
password.<BR>;<BR><BR>exten => 2999,1,VoicemailMain(${CALLERIDNUM})</FONT></DIV> <DIV><FONT color=#003366></FONT> </DIV> <DIV>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 :</DIV> <DIV> </DIV> <DIV>pbx.c:1731: can not find extension context 'from-sip'</DIV> <DIV> </DIV> <DIV>when i close asterisk server communication is taking place beween both phones.</DIV> <DIV>now i m stuck with this error. can anybody give me guidance on how to solve this problem?</DIV> <DIV> </DIV> <DIV>thanks</DIV> <DIV> </DIV> <DIV>tejas</DIV></DIV></DIV><p>
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