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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ho Chan Sent: Jueves, 24 de Febrero de 2005 09:26 p.m. To: [email protected] 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) _________________________________________________________________ Get 10Mb extra storage for MSN Hotmail. Subscribe Now! http://join.msn.com/?pgmarket=en-hk _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
