Hi Marc,<div><br></div><div>As you can see from your trace, the call is never completely established, as the 200 OK's ACK never makes it back to the user-agent.</div><div>After a brief look this seems to be caused by the gateway always using the contact header to respond instead of the from header, as for the BYE message also being rejected for the same reason - just my 2 cents, might be wrong - maybe someone else on the list will have different input,</div>
<div><br></div><div>Best,</div><div><br></div><div><div style="font-family:Arial;font-size:13.200000762939453px;background-color:rgb(255,255,255);color:rgb(119,119,119)">Lorenzo Mangani</div><div style="font-family:Arial;font-size:13.200000762939453px;background-color:rgb(255,255,255);color:rgb(119,119,119)">
<div style="font-size:x-small"><br></div><div style="font-size:x-small">HOMER DEV TEAM</div></div><div style="font-family:Arial;background-color:rgb(255,255,255);color:rgb(119,119,119);font-size:x-small"><font size="1">QXIP - Network Engineering</font></div>
</div><div><br></div><div><br><br><div class="gmail_quote">On Sun, Mar 24, 2013 at 4:54 PM, Marc Rys <span dir="ltr"><<a href="mailto:m.rys@tri-lakes.net" target="_blank">m.rys@tri-lakes.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
ok I'll stop changing the subject, even though I regret not naming it SPCE Taqua Integration..<br>
<br>
Anyways, I've heeded your advice and I normalized the patterns and I'm now routing the calls from my Taqua PSTN gateway to my IP phones. Now the newest problem to reveal itself is the incoming calls to my Phones fail to setup completely. It appears all the call routing is working properly now, I can call my DID from the PSTN, and my IP phone rings, but when I pickup the call, I hear no audio, and it appears the phone never completely sets up the call. The screen still shows the phone as ringing.<br>
<br>
The wireshark cap shows RTP moving between my gateway & SPCE, and RTP moving between SPCE and my IP phone, but the call never appears to setup completely on the IP phone.<br>
<br>
Any thoughts?<br>
<div class="im HOEnZb"><br>
Marc Rys<br>
<a href="http://www.tri-lakes.net" target="_blank">http://www.tri-lakes.net</a><br>
<a href="http://www.rystec.com" target="_blank">http://www.rystec.com</a><br>
<br>
<br>
</div><div class="HOEnZb"><div class="h5">----- Original Message -----<br>
From: "Lorenzo Mangani" <<a href="mailto:lorenzo.mangani@gmail.com">lorenzo.mangani@gmail.com</a>><br>
To: "Marc Rys" <<a href="mailto:m.rys@tri-lakes.net">m.rys@tri-lakes.net</a>><br>
Cc: <a href="mailto:spce-user@lists.sipwise.com">spce-user@lists.sipwise.com</a><br>
Sent: Saturday, March 23, 2013 4:39:50 PM<br>
Subject: Re: [Spce-user] 483 Too Many Hops<br>
<br>
Marc,<br>
<br>
<br>
Please don't fork the messages by creating a new thread for each step of the discussion and consult the documentation.<br>
You need an inbound rewrite rule applied to strip the + from the INVITE in order to match the local user, as well described in the Handbook, actually this is exactly the example shown there: <a href="http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans" target="_blank">http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans</a><br>
<br>
<br>
<br>
Lorenzo Mangani<br>
<br>
<br>
<br>
HOMER DEV TEAM<br>
QXIP - Network Engineering<br>
<br>
<br>
On Sat, Mar 23, 2013 at 9:55 PM, Lorenzo Mangani < <a href="mailto:lorenzo.mangani@gmail.com">lorenzo.mangani@gmail.com</a> > wrote:<br>
<br>
<br>
Marc,<br>
<br>
<br>
Your Tarqua invite has Max-Forwards set to 1.<br>
Try increasing the allowed hops and the call will terminate.<br>
<br>
<br>
<br>
<br>
<br>
Lorenzo Mangani<br>
<br>
<br>
<br>
HOMER DEV TEAM<br>
QXIP - Network Engineering<br>
<br>
<br>
<br>
On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys < <a href="mailto:m.rys@tri-lakes.net">m.rys@tri-lakes.net</a> > wrote:<br>
<br>
<br>
<br>
<br>
I've been evaluating SPCE over the last week. I've already setup a couple test subscribers and setup a peer with a provider we work with for SIP LD Term. All of those test have worked very successful.<br>
<br>
We also have our own media gateway which is interconnected with the local PSTN via TDM trunk, but I send incoming calls from the PSTN through our MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops". Attached is the wireshark cap.<br>
<br>
Any help is appreciated.<br>
<br>
Marc Rys<br>
<a href="http://www.tri-lakes.net" target="_blank">http://www.tri-lakes.net</a><br>
<a href="http://www.rystec.com" target="_blank">http://www.rystec.com</a><br>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br><span style="font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)"><span style="font-family:Arial"><div style="color:rgb(119,119,119)">
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