Forgot to post the response to the list as well.<br><br><div class="gmail_quote">Date: Fri, Jun 22, 2012 at 6:57 AM<br>Subject: Re: [SR-Users] Can Kamailio be used to redirect media between a client that switches from wifi to 3g/gsm<br>
To: Klaus Darilion <<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>><br><br><br>Thanks for the response! I see a series of what I believe are re-REGISTER statements:<div><br></div><div>
<div><div>Message sent: (to dest=75.101.244.XXX:5060)</div><div>REGISTER sip:75.101.244.XXX SIP/2.0</div><div>Via: SIP/2.0/UDP 10.165.27.161:2407;rport;branch=z9hG4bK1839704852</div>
<div>From: <sip:990XX@75.101.244.XXX>;tag=1689684502</div><div>To: <sip:990XX@75.101.244.XXX></div><div>Call-ID: 1867622191</div><div>CSeq: 1 REGISTER</div><div>Contact: <sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22></div>
<div>Max-Forwards: 70</div><div>User-Agent: Linphone/3.4.0 (eXosip2/unknown)</div><div>Expires: 3600</div><div>Content-Length: 0</div><div><br></div><div>Received message:</div><div>SIP/2.0 401 Unauthorized</div><div>Via: SIP/2.0/UDP 10.165.27.161:2407;received=<a href="tel:32.158.143.61" value="+13215814361" target="_blank">32.158.143.61</a>;rport=2407;branch=z9hG4bK1839704852</div>
<div>From: <sip:990XX@75.101.244.XXX>;tag=1689684502</div><div>To: <sip:990XX@75.101.244.XXX>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.e10d</div><div>Call-ID: 1867622191</div><div>CSeq: 1 REGISTER</div><div>WWW-Authenticate: Digest realm="75.101.244.XXX", nonce="4fe476e500000e00aa8700d49b8c668f4c6f1e6a367f2XXX"</div>
<div>Server: Kamailio</div><div>Content-Length: 0</div><div><br></div><div>REGISTER sip:75.101.244.XXX SIP/2.0</div><div>Via: SIP/2.0/UDP 10.165.27.161:2407;rport;branch=z9hG4bK123406454</div><div>From: <sip:990XX@75.101.244.XXX>;tag=1689684502</div>
<div>To: <sip:990XX@75.101.244.XXX></div><div>Call-ID: 1867622191</div><div>CSeq: 2 REGISTER</div><div>Contact: <sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22></div><div>Authorization: Digest username="990XX", realm="75.101.244.XXX", nonce="4fe476e500000e00aa8700d49b8c668f4c6f1e6a367f2XXX", uri="sip:75.101.244.XXX", response="1e1d558894f2c05c322c76efbb2f9XXX", algorithm=MD5</div>
<div>Max-Forwards: 70</div><div>User-Agent: Linphone/3.4.0 (eXosip2/unknown)</div><div>Expires: 3600</div><div>Content-Length: 0</div><div><br></div><div>Received message:</div><div>SIP/2.0 200 OK</div><div>Via: SIP/2.0/UDP 10.165.27.161:2407;received=<a href="tel:32.158.143.61" value="+13215814361" target="_blank">32.158.143.61</a>;rport=2407;branch=z9hG4bK123406454</div>
<div>From: <sip:990XX@75.101.244.XXX>;tag=1689684502</div><div>To: <sip:990XX@75.101.244.XXX>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.8b5a</div><div>Call-ID: 1867622191</div><div>CSeq: 2 REGISTER</div><div>Contact: <sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22>;expires=120, <sip:990XX@50.43.101.83:51879;line=59ecc207f06f4e9>;expires=81</div>
<div>Server: Kamailio</div><div>Content-Length: 0</div></div><div><br></div><div>But after this I would expect to see an INVITE but one is never sent, but if I switch back to the original IP on that device the call is reconnected, so it proves we're missing an INVITE I believe. What do I need to do on the server side to force a re-INVITE to be sent after this registration occurs? Thanks!</div>
<div><div class="h5">
<br><div class="gmail_quote">On Fri, Jun 22, 2012 at 1:14 AM, Klaus Darilion <span dir="ltr"><<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@pernau.at</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Shaun!<br>
<br>
Your problem description is too short to give you any good help.<br>
<br>
Use tcpdump (or other tools) to capture the scenario with Asterisk and Kamailio. Then compare them to find out why it doesn't work.<br>
<br>
Is media sent directly to Asterisk then it ca not be the problem of Kamailio.<br>
<br>
I hope the mobile client is smart enough to also send a reINVITE when getting the new IP address (of the mobile connection) with proper Contact header - otherwise it can not receive SIP requests from Asterisk.<br>
<br>
regards<br>
Klaus<div><div><br>
<br>
On 20.06.2012 18:07, Shaun Clark wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>
The use case is that I have a SIP client registered to Kamailio talking<br>
to an Asterisk box connected to the PSTN. The client is a mobile phone<br>
and the user is connected to wifi. The user then steps out of wifi range<br>
and the phone drops the connection and picks up the 3g data connection.<br>
I want the media stream to reconnect to the client and the call to<br>
resume without having to redial. This works now if the client is<br>
directly connected to the Asterisk machine, but not when I am routing<br>
through my Kamailio server. How do I go about this, examples are always<br>
appreciated, thanks!<br>
<br>
<br></div></div>
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</blockquote>
<br>
<br>
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