Crazy Boy
2006-Jul-27 05:36 UTC
[asterisk-users] Problem with call receiving (Asterisk+PSTN+Digium TDM04B)
Hi Friends, I am Chandra from India. Thank you for your cooperation and for clear my doubts. Now, I have installed Digium TDM04B card in my Asterisk server and configured. I have one landline number from PSTN. Now, I have connected that PSTN cable to my TDM04B first port. When I am making calls from outside to my PSTN number, sometimes Asterisk receving that call and sometimes, its not receiving. Why? Here I am giving my configuration of my files. ZAPTEL.CONF contents: loadzone = us defaultzone=us fxsks=1,2,3,4 ZAPATA.CONF contents: [channels] context=tutorial signalling=fxs_ks busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callerid=asreceived language=en usecallerid=yes echocancel=yes transfer=yes immediate=no group=1 channel => 1 SIP.CONF contents: [300] type=friend username=300 secret=server callerid="Server" host=dynamic context=tutorial [general] port=5060 bindaddr=0.0.0.0 context=default disallow=all allow=gsm allow=ulaw allow=alaw EXTENSIONS.CONF contents: TRUNK=Zap/g1 TRUNK=Zap/g2 [tutorial] exten => s,1,Dial(SIP/350,30) exten => s,n,Voicemail(350) exten => s,n,Hangup exten => 300,1,Dial(SIP/300,15) exten => 300,2,Voicemail(u300) exten => 300,3,Voicemail(b300) exten => 300,4,Hangup What is the solution? Please tell me. Looking forward to your response. Thank you. Regards, Chandra. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060727/7d1507b2/attachment.htm
Filip Drągowski
2006-Jul-27 05:51 UTC
[asterisk-users] Problem with call receiving (Asterisk+PSTN+Digium TDM04B)
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-2" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> First<br> zaptel.conf<br> fxsks=1-4<br> zapata.conf <br> channel => 1-4<br> extensions.conf<br> [tutorial]<br> exten => s,1,Dial(SIP/350,30)<br> - do You have SIP/350 ? ther is onlu 300 in sip.conf<br> <br> <br> <blockquote cite="mid20060727123611.6899.qmail@web37101.mail.mud.yahoo.com" type="cite"><tt>Hi Friends, <br> <br> I am Chandra from India. Thank you for your cooperation and for clear my doubts. <br> <br> Now, I have installed Digium TDM04B card in my Asterisk server and configured. I have one landline number from PSTN. Now, I have connected that PSTN cable to my TDM04B first port. <span style="font-weight: bold;">When I am making calls from outside to my PSTN number, sometimes Asterisk receving that call and sometimes, its not receiving. Why?</span> Here I am giving my configuration of my files. <br> <br> <span style="font-weight: bold;">ZAPTEL.CONF contents:</span> <br> <br> loadzone = us<br> defaultzone=us<br> fxsks=1,2,3,4 <br> <br> <span style="font-weight: bold;">ZAPATA.CONF contents:</span> <br> <br> [channels]<br> context=tutorial<br> signalling=fxs_ks<br> busydetect=1<br> busycount=7<br> relaxdtmf=yes<br> callwaiting=yes<br> callwaitingcallerid=yes<br> threewaycalling=yes<br> cancallforward=yes<br> echocancelwhenbridged=yes<br> rxgain=0.0<br> txgain=0.0</tt> <pre> callerid=asreceived</pre> <pre><tt> language=en</tt></pre> <pre><tt> usecallerid=yes</tt></pre> <pre><tt> echocancel=yes</tt></pre> <pre><tt> transfer=yes</tt></pre> <pre><tt> immediate=no</tt></pre> <pre><tt> group=1</tt></pre> <pre><tt> channel => 1</tt></pre> <pre><tt> <span style="font-weight: bold;">SIP.CONF contents:</span></tt></pre> <pre><tt> [300]</tt></pre> <pre><tt> type=friend</tt></pre> <pre><tt> username=300</tt></pre> <pre><tt> secret=server</tt></pre> <pre><tt> callerid="Server"</tt></pre> <pre><tt> host=dynamic</tt></pre> <pre><tt> context=tutorial</tt></pre> <pre><tt> [general]</tt></pre> <pre><tt> port=5060 </tt></pre> <pre><tt> bindaddr=0.0.0.0</tt></pre> <pre><tt> context=default</tt></pre> <pre><tt> disallow=all</tt></pre> <pre><tt> allow=gsm</tt></pre> <pre><tt> allow=ulaw</tt></pre> <pre><tt> allow=alaw</tt></pre> <pre><tt> <span style="font-weight: bold;">EXTENSIONS.CONF contents:</span></tt></pre> <pre><tt> TRUNK=Zap/g1</tt></pre> <pre><tt> TRUNK=Zap/g2 </tt></pre> <pre><tt> [tutorial]</tt></pre> <pre><tt> exten => s,1,Dial(SIP/350,30)</tt></pre> <pre><tt> exten => s,n,Voicemail(350)</tt></pre> <pre><tt> exten => s,n,Hangup</tt></pre> <pre><tt> exten => 300,1,Dial(SIP/300,15)</tt></pre> <pre><tt> exten => 300,2,Voicemail(u300)</tt></pre> <pre><tt> exten => 300,3,Voicemail(b300)</tt></pre> <pre><tt> exten => 300,4,Hangup</tt></pre> <pre><tt> What is the solution? Please tell me. Looking forward to your response. </tt></pre> <pre><tt> Thank you.</tt></pre> <pre><tt> Regards,</tt></pre> <pre><tt> Chandra.</tt></pre> </blockquote> <br> </body> </html>