Hi,<br><br>Please read bellow:<br><br><div><span class="gmail_quote">On 9/1/06, <b class="gmail_sendername">Crazy Boy</b> <<a href="mailto:crazymoonboy@yahoo.com">crazymoonboy@yahoo.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>Hi friends,<br> <br>Thank you to all for your response and cooperation to me. I have a doubt.<br><br>We have registered with Teliax and got DID number. We are making calls to USA successfully using your service. But, We are unable to receive incoming calls to our DID. Here I am sending my config files and error message on Asterisk console.
<br> <br> <span style="font-weight: bold;"> Contents in IAX.CONF file:</span><br> <br> disallow=all <br> allow = ulaw <br> <br> [general]<br> register => <a href="mailto:teliaxusername:teliaxpassword@voip-co1.teliax.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
teliaxusername:teliaxpassword@voip-co1.teliax.com</a> <br> <br> [teliax]<br> context=telincoming<br> type=friend<br> host=<a href="http://voip-co1.teliax.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
voip-co1.teliax.com</a><br> auth=md5<br> secret=teliaxpassword<br> disallow=all<br> allow=ulaw<br> allow=alaw<br> allow=gsm<br>
<br> <span style="font-weight: bold;"><span style="font-weight: bold;">Contents in Sip.conf file:<br> <br> </span></span>[105]<br> type=friend<br> username=105<br> secret=ravi<br> callerid="RaviKanth"<br> host=dynamic
<br> context=leader<br> canreinvite=no<br> nat=yes<br> dtmfmode=rfc2833<br> allow=all<br> mailbox=605@vmail<br> <br> [107]<br> type=friend<br> username=107<br> secret=suresh<br> callerid="Suresh"<br> host=dynamic
<br> context=administration<br> canreinvite=no<br> nat=yes<br> dtmfmode=rfc2833<br> allow=all<br> mailbox=607@vmail<span style="font-weight: bold;"><span style="font-weight: bold;"><span style="font-weight: bold;"><br> </span>
<br> </span><span style="font-weight: bold;"></span>Contents in Extensions.conf file:</span><br> <br> [telincoming]<br> exten => 303xxxxxxx, 1, Answer()<br> exten => 303xxxxxxx, n, Wait,2<br> exten => 303xxxxxxx, n, Goto(incoming,s,1)
</div></blockquote><div><br>You need to inser "_" before a pattern so asterisk can try to match it:<br> exten => _303xxxxxxx, 1, Answer()<br>exten => _303xxxxxxx, n, Wait,2<br> exten => _303xxxxxxx, n, Goto(incoming,s,1)
<br><br>Should solve your problem!<br><br>Also only as debug you can try _X. <br><br>Pls tell me if it solved your problem.<br><br></div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div> include => internal<br> include => incoming<br> <br> [incoming]<br>
exten => s,1,Wait(3)<br> exten => s,n,Answer<br> exten => s,n,SetMusicOnHold(default)<br> exten => s,n,Set(TIMEOUT(digit)=5)<br> exten => s,n,Set(TIMEOUT(response)=10)<br> exten => s,n,Background(/tmp/virg2)
<br> exten => s,n,Goto(s,1)<br> exten => s,n,Hangup()<br> include => internal<br> <br> [internal]<br> exten => 105,1,SetMusicOnHold(default)<br> exten => 105,2,Dial(SIP/105,7,t,m,T)<br> exten => 1605,1,VoiceMailMain(
605@vmail)<br> exten => 105,3,VoiceMail(605@vmail)<br> exten => 105,4,Hangup<br> <br> exten => 107,1,SetMusicOnHold(default)<br> exten => 107,2,Dial(SIP/107,7,t,m,T)<br> exten => 1607,1,VoiceMailMain(
607@vmail)<br> exten => 107,3,VoiceMail(607@vmail)<br> exten => 107,4,Hangup<br> <br> [uscall]<br> exten => _1XXXXXXXXXX,1,DIAL(IAX2/teliaxusername@teliax/${EXTEN},30,tr) <br> <br> [manager]<br> include => uscall
<br> include => internal<br> <br> <span style="font-weight: bold;">The error message on Asterisk console:</span><br> <br> *CLI> -- Executing Dial("SIP/105-007951e0", "IAX2/teliaxusername@teliax
/1303xxxxxxx|30|tr") in new stack<br> -- Called teliaxusername@teliax/1303xxxxxxx<br> -- Call accepted by <a href="http://207.174.202.2/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
207.174.202.2</a> (format ulaw)<br> -- Format for call is ulaw<br> -- IAX2/teliax-1 is ringing<br> -- IAX2/teliax-1 is making progress passing it to SIP/105-007951e0<br> -- IAX2/teliax-1 is ringing
<br> -- IAX2/teliax-1 is busy<br> -- Hungup 'IAX2/teliax-1'<br> == Everyone is busy/congested at this time (1:1/0/0)<br> == Auto fallthrough, channel 'SIP/105-007951e0' status is 'BUSY'<br> <br> <br> What is the
problem? Can you please tell me the solution. Looking forward to your response. Thank you.<br> <br> Regards,<br> Chandra.<br></div><div><span class="ad"> <p>
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