After doing some reading, thinking that I may be able to solve my problem using the dialog module and fix_route_dialog(). Since the Contact Header is being re-written by the Provider, will fix_route_dialog() change it on the way in to the appropriate value in the dialog, causing the message to be forwarded correctly? The code for what I'm thinking is below. Unfortunately, the issue is so intermittent that I have not been able to re-produce the case of the Provider re-writing the Contact Header since I started trying this modification. Does anyone know if this strategy will work?<div>
<br></div><div><div> if (!has_totag()) {</div><div> create_dialog();</div><div> # initial request</div><div> record_route();</div><div> }</div><div> else {</div>
<div> # sequential request - obey the indicated route</div><div> loose_route();</div><div><br></div><div> if ($DLG_status!=NULL) {</div><div> if (!validate_dialog()) {</div>
<div> fix_route_dialog();</div><div> }</div><div> }</div><div><br></div><div> t_relay();</div><div> exit;</div><div> }</div><div><br>
</div><br><div class="gmail_quote">On Fri, Nov 11, 2011 at 9:23 AM, Matt Stockton <span dir="ltr"><<a href="mailto:mstockton@harqen.com">mstockton@harqen.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
I added the following to my config:<div><br></div><div><div>onreply_route[pstn_outgoing] {</div><div class="im"><div> setflag(1);</div><div> if(t_check_status("200")) {</div><div> if(search("Content-type: application/sdp")) {</div>
<div> fix_nated_sdp("8", "x.x.x.x");</div><div> }</div><div> }</div><div>}</div></div></div><div><br></div><div>And the only thing that seemed to have changed in the SIP signalling is, when my provider sends SIP 200 OK to OpenSips, OpenSips is just re-writing the OK Content before sending onto Freeswitch. It doesn't look like any messages have changed that are going from OpenSips to the provider, and the provider is still sending the BYE message with the Contact Header intermittently to the different IPs as I described. Any more thoughts on this? </div>
<div><br></div><div>My end goal is to simply have the same OpenSips act as both an inbound proxy and outbound proxy, with no registration requirements. If there is an easier way to do this with OpenSips, please let me know. I've looked around and checked out all the referenced tutorials on the site w/r/t load balancing and routing</div>
<div><br></div><div><div><span style="white-space:pre-wrap"> </span>v=0</div><div><span style="white-space:pre-wrap"> </span>o=Sonus_UAC 14270 3732 IN IP4 x.x.x.x</div><div><span style="white-space:pre-wrap"> </span>s=SIP Media Capabilities</div>
<div><span style="white-space:pre-wrap"> </span>c=IN IP4 67.231.0.110</div><div><span style="white-space:pre-wrap"> </span>t=0 0</div><div><span style="white-space:pre-wrap"> </span>m=audio 25620 RTP/AVP 0 101</div>
<div><span style="white-space:pre-wrap"> </span>a=rtpmap:0 PCMU/8000</div><div><span style="white-space:pre-wrap"> </span>a=rtpmap:101 telephone-event/8000</div><div><span style="white-space:pre-wrap"> </span>a=fmtp:101 0-15</div>
<div><span style="white-space:pre-wrap"> </span>a=sendrecv</div><div><span style="white-space:pre-wrap"> </span>a=maxptime:20</div><div><span style="white-space:pre-wrap"> </span>a=oldmediaip:ACTUAL_IP</div>
</div><div class="HOEnZb"><div class="h5"><div><br></div><div><br><br><div class="gmail_quote">On Thu, Nov 10, 2011 at 7:17 PM, ddgiants <span dir="ltr"><<a href="mailto:ddgiants@gmail.com" target="_blank">ddgiants@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Matt,<br>
Sounds like you may not be topology hiding the 200OKs. Some/most SIP<br>
devices/SBC's etc will send a BYE to the ip address in the o(owner) line of<br>
the SDP of the 200OK. Some devices do not and send BYE to the initator/proxy<br>
they are configured for. The 200OK will have o and c. O is the owner,<br>
meaning signaling owner and C is the connection IP as in where to send the<br>
actual RTP. I use the below to change the o portion of the SDP in 200 OKs<br>
and works for me. Give it a try and let me know.<br>
<br>
onreply_route[1] {<br>
setflag(1);<br>
if(t_check_status("200")) {<br>
if(search("Content-type: application/sdp")) {<br>
fix_nated_sdp("8", "x.x.x.x");<br>
}<br>
}<br>
}<br>
<br>
DD<br>
<span><font color="#888888"><br>
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