Seeing as you have posted this to -biz I am assuming you want to pay for this to get fixed.<br>
<br>
Post this message to -users and you might be able to get this fixed for free.<br>
<br>
that'll be $100 please.<br>
<br>
Mark.<br><br><div><span class="gmail_quote">On 9/25/05, <b class="gmail_sendername">jonny hashem</b> &lt;<a href="mailto:jonnyhashem@yahoo.com">jonnyhashem@yahoo.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
i have an asterisk box (<a href="http://195.112.214.99">195.112.214.99</a>) with this<br>configuration:<br><br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;sip.conf<br>[callshop]<br>type=peer<br>host=<a href="http://sip.callshopcompany.com">sip.callshopcompany.com
</a><br>username=XXXXXXX<br>secret=XXXXXX<br>allow=all<br><br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;extensions.conf<br><br>[call]<br>exten =&gt; _00.,1,Dial,SIP/callshop/${EXTEN}<br><br>and when i try to send calls to the voip provider<br>
(callshopcompany &quot;<a href="http://213.61.187.150">213.61.187.150</a>&quot;) i got these<br>messages:<br><br>*CLI&gt; dial 0017046872001@call<br>&nbsp;&nbsp;&nbsp;&nbsp;-- Executing Dial(&quot;OSS/dsp&quot;,<br>&quot;SIP/callshop/0017046872001&quot;) in new stack
<br>&nbsp;&nbsp;&nbsp;&nbsp;-- Called callshop/0017046872001<br>*CLI&gt; Sep 24 14:16:45 WARNING[22295]: chan_sip.c:6890<br>handle_response: Forbidden - wrong password on<br>authentication for INVITE to '&quot;asterisk&quot;<br>&lt;sip:asterisk@195.112.214.99
:5070&gt;;tag=as4cda63c2'<br>&nbsp;&nbsp;&nbsp;&nbsp;-- SIP/callshop-f613 is circuit-busy<br>&nbsp;&nbsp;== Everyone is busy/congested at this time<br>&nbsp;&nbsp;&nbsp;&nbsp;-- Got SIP response 481 &quot;Call Leg Does Not Exist&quot;<br>back from <a href="http://213.61.187.150">
213.61.187.150</a><br>Sep 24 14:16:58 WARNING[22295]: pbx.c:1949<br>ast_pbx_run: Timeout, but no rule 't' in context<br>'call'<br> &lt;&lt; Hangup on console &gt;&gt;<br><br>but when ive tried it on xlite in the same<br>configuration to send calls to the same company it
<br>worked and the calls passed without any problems.<br><br>so whats the problem here,why the call goes well using<br>xlite and fails using asterisk despite they have the<br>same configuration.<br><br>__________________________________________________
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</div><br><br clear="all"><br>-- <br>regards,<br><br>Mark P. Edwards<br>FWD: 667917<br>