<div>Thank you for your response.</div> <div> </div> <div>All that I get when I dial in is all circutes are busy and when I dial out 503 errors. Here are my configs. Any ideas would be greatly appreciated. The provider is using a Tekelec 9000 Class 5 switch if that is any help.</div> <div> </div> <div>The provider sat us up two accounts one that they are calling a line that uses authentication usernames and passwords as though we were terminating into a SIP phone. This account works but is not the prefered solution to our problem. The other account is what the provider calls a trunk account and it does not use usernames and passwords. This seems to be the solution for the provider and us.</div> <div> </div> <div> </div> <div>Thank you,</div> <div>Steven</div> <div> </div> <div> </div> <div>sip.conf</div> <div> </div>
<div>[inbound-trunk]<BR>type=friend<BR>context=incoming<BR>insecure=very<BR>host=xxx.xxx.xxx.xxx<BR>outboundproxy=xxx.xxx.xxx.xxx<BR>fromdomain=xxx.xxx.xxx.xxx<BR>defaultip=xxx.xxx.xxx.xxx<BR>disallow=all<BR>allow=ulaw<BR>nat=yes<BR>canreinvite=yes<BR>qualify=yes<BR></div> <div>extension.conf</div> <div> </div> <div>[incoming]<BR>exten => NXXNXXXXXX,1,Answer()<BR>exten => NXXNXXXXXX,2,Background(greeting)<BR>exten => NXXNXXXXXX,3,SayDigits(${CALLERIDNUM})<BR>exten => NXXNXXXXXX,4,Dial(SIP/steven)<BR>exten => NXXNXXXXXX,5,Hangup()<BR><BR><BR><B><I>C F <shmaltz@gmail.com></I></B> wrote:</div> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">It should work as is, just make usre that you have an extension<BR>defined (or a catch all) for every DID you have with the provider so<BR>that incoming works.<BR><BR>On 6/2/06, Steven Haldeman <HALDEMAN_79@YAHOO.COM>wrote:<BR>><BR>>
Hello,<BR>><BR>> I am attempting to figure out how to set up SIP trunking, between my company<BR>> and our SIP provider. This is an expermintal project at this time. The SIP<BR>> provider gave us a Signalling IP address and two Media IP addresses. We<BR>> supplied them with the IP address of our Asterisk box. When asked what our<BR>> Usernames and Passwords would be we were told that they were not needed for<BR>> a SIP trunk. We can use what they call SIP lines that use username/password<BR>> however because of tarrifing the lines cost more per month than a trunk I<BR>> have been successfull in making a SIP Line work, but have no idea where to<BR>> start with a SIP Trunk. We sill be using DID numbers on the SIP trunks.<BR>> Has anyone had any experiance with this type of configuration an example<BR>> would be extrememly helpfull.<BR>><BR>> I have search the internet for help and I may have seen a solution but just<BR>> was not
certain what I was looking at, or how to implement as everything<BR>> that I have seen user a Username/Password combo.<BR>><BR>> Thank you in advance.<BR>><BR>> Thanks<BR>><BR>> Steven<BR>><BR>><BR>> __________________________________________________<BR>> Do You Yahoo!?<BR>> Tired of spam? Yahoo! Mail has the best spam protection around<BR>> http://mail.yahoo.com<BR>> _______________________________________________<BR>> --Bandwidth and Colocation provided by Easynews.com --<BR>><BR>> Asterisk-Users mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>><BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>><BR>><BR>><BR>_______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>Asterisk-Users mailing list<BR>To UNSUBSCRIBE or update options
visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE><BR><p> __________________________________________________<br>Do You Yahoo!?<br>Tired of spam? Yahoo! Mail has the best spam protection around <br>http://mail.yahoo.com