<div dir="ltr">Thanks a lot Michael.I am a real begginer in the SIP world and your response was very helpful. It worked :))) <br>Thanks again.<br><div class="gmail_quote">2012/4/16 Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br><br><div class="gmail_quote"><div class="im">On Mon, Apr 16, 2012 at 8:36 AM, azza miled <span dir="ltr"><<a href="mailto:azza.miled@gmail.com" target="_blank">azza.miled@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><span>Thanks for the response, but evec with these steps, I still have the error "call failed:proxy authentication required" when I dial the IVR extension from an opensips' phone. Do any one know, what shall I do???</span> <div>
<div><br></div></div></div></blockquote></div>Do you need SIP authentication between your FS box and your OpenSIPS box? I'm guessing not. If that's the case then I recommend opening up this file:<br><br>/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml<br>
<br>Find the tag with this opening line:<br><list name="domains" default="deny"><br><br>Add a new line in there like this:<br><node type="allow" cidr="x.x.x.x/32"/><br>
where x.x.x.x is the IP address of your OpenSIPS server.<br>
<br>then go to fs_cli and issue this command: reloadacl<br><br>Lastly, you'll need to route the inbound calls from OpenSIPS through your 'public' dialplan context because that's where ACL-allowed inbound calls normally get processed. You can either create your IVRs in the public context or you can transfer the inbound call over to the default context (or whatever context you are using.) <br>
<span class="HOEnZb"><font color="#888888">
<br>-MC<br></font></span></div>
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