Hi,<br><br>Currently I&#39;m working on case i.e. OpenSips and Asterisk, where I&#39;m using OpenSIps as a Proxy server using dispatcher module and dispatcher list contains Asterisk machines IP as destination address.<br><br>

The configuration I&#39;ve done in opensIps.cfg is listed down below;<br><br>        if (is_method(&quot;INVITE&quot;)) {<br>                ds_select_dst(&quot;1&quot;, &quot;4&quot;);<br>                forward();<br>                route(1);<br>

                setflag(1); # do accounting<br>        }<br><br><br>My UAC IP is xx.xx.xx.xx, OpenSips IP: yy.yy.yy.yy and Asterisk IP: zz.zz.zz.zz.<br><br>When I make a call I&#39;m getting code error 603, Decline. Even though the settings I&#39;ve set on UAC as outbound proxy using valid credentials as used on my Asterisk machine.<br>

<br>Kindly advise me how can I send Registration request OpenSips -&gt; Asterisk. Please give me some sample to resolve this issue. At the end I&#39;m listing few traces;<br><br>U zz.zz.zz.zz:5060 -&gt; yy.yy.yy.yy:5060<br>
SIP/2.0 401 Unauthorized.<br>Via: SIP/2.0/UDP 77.66.2.137;branch=z9hG4bK-d87543-928337242-1--d87543-;received=yy.yy.yy.yy.<br>Via: SIP/2.0/UDP 192.168.0.168:5060;received=203.215.176.22;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183.<br>
From: 3225555025&lt;sip:3225555025@yy.yy.yy.yy.&gt;;tag=9d7c6756.<br>To: 3225555025&lt;sip:3225555025@yy.yy.yy.yy.&gt;;tag=as0f0e0e90.<br>Call-ID: b115ce088a57d010.<br>CSeq: 8160 REGISTER.<br>User-Agent: Asterisk PBX.<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<br>Supported: replaces.<br>WWW-Authenticate: Digest algorithm=MD5, realm=&quot;<a href="http://rtsip.vopium.com">rtsip.vopium.com</a>&quot;, nonce=&quot;0a26e4a7&quot;, stale=true.<br>
Content-Length: 0.<br>.<br><br><br>U yy.yy.yy.yy.:5060 -&gt; xx.xx.xx.xx:46183<br>SIP/2.0 401 Unauthorized.<br>Via: SIP/2.0/UDP 192.168.0.168:5060;received=xx.xx.xx.xx;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183.<br>
From: 3225555025&lt;sip:3225555025@yy.yy.yy.yy.&gt;;tag=9d7c6756.<br>To: 3225555025&lt;sip:3225555025@yy.yy.yy.yy.&gt;;tag=as0f0e0e90.<br>Call-ID: b115ce088a57d010.<br>CSeq: 8160 REGISTER.<br>User-Agent: Asterisk PBX.<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<br>Supported: replaces.<br>WWW-Authenticate: Digest algorithm=MD5, realm=&quot;<a href="http://rtsip.vopium.com">rtsip.vopium.com</a>&quot;, nonce=&quot;0a26e4a7&quot;, stale=true.<br>
Content-Length: 0.<br><br><br>U zz.zz.zz.zz:5060 -&gt; yy.yy.yy.yy:5060<br>SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP 77.66.2.137;branch=z9hG4bK-d87543-928337242-1--d87543-;received=77.66.2.137.<br>Via: SIP/2.0/UDP 77.66.2.137;branch=z9hG4bK-d87543-928337242-1--d87543-.<br>
Via: SIP/2.0/UDP 192.168.0.168:5060;received=203.215.176.22;received=203.215.176.22;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183;rport=46183.<br>From: 3225555025&lt;sip:3225555025@yy.yy.yy.yy&gt;;tag=9d7c6756.<br>
To: 3225555025&lt;sip:3225555025@yy.yy.yy.yy&gt;.<br>Call-ID: b115ce088a57d010.<br>CSeq: 8160 REGISTER.<br>User-Agent: Asterisk PBX.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<br>Supported: replaces.<br>
Contact: &lt;sip:3225555025@zz.zz.zz.zz&gt;.<br>Content-Length: 0.<br>.<br><br><br>U zz.zz.zz.zz:5060 -&gt; yy.yy.yy.yy:5060<br>SIP/2.0 401 Unauthorized.<br>Via: SIP/2.0/UDP 77.66.2.137;branch=z9hG4bK-d87543-928337242-1--d87543-;received=77.66.2.137.<br>
Via: SIP/2.0/UDP 77.66.2.137;branch=z9hG4bK-d87543-928337242-1--d87543-.<br>Via: SIP/2.0/UDP 192.168.0.168:5060;received=203.215.176.22;received=203.215.176.22;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183;rport=46183.<br>
From: 3225555025&lt;sip:3225555025@yy.yy.yy.yy&gt;;tag=9d7c6756.<br>To: 3225555025&lt;sip:3225555025@yy.yy.yy.yy&gt;;tag=as0f0e0e90.<br>Call-ID: b115ce088a57d010.<br>CSeq: 8160 REGISTER.<br>User-Agent: Asterisk PBX.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<br>
Supported: replaces.<br>WWW-Authenticate: Digest algorithm=MD5, realm=&quot;<a href="http://rtsip.vopium.com">rtsip.vopium.com</a>&quot;, nonce=&quot;2c6529c9&quot;, stale=true.<br>Content-Length: 0.<br>.<br><br><br>U yy.yy.yy.yy:5060 -&gt; xx.xx.xx.xx:46183<br>
SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP 192.168.0.168:5060;received=203.215.176.22;received=203.215.176.22;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183;rport=46183.<br>From: 3225555025&lt;sip:3225555025@yy.yy.yy.yy&gt;;tag=9d7c6756.<br>
To: 3225555025&lt;sip:3225555025@yy.yy.yy.yy&gt;.<br>Call-ID: b115ce088a57d010.<br>CSeq: 8160 REGISTER.<br>User-Agent: Asterisk PBX.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<br>Supported: replaces.<br>
Contact: &lt;sip:3225555025@zz.zz.zz.zz.zz&gt;.<br>Content-Length: 0.<br>.<br><br><br clear="all"><br>-- <br>Regards,<br>
<br>Ahmed Munir<br><br><br>