Shlomo,<br>My understanding is I have to do a no fixup sip 5060. This from Cisco. Without doing the no fixup the registration ports get all mangled.<br><br><div class="gmail_quote">On Nov 27, 2007 10:11 AM, Shlomo Dubrowin <
<a href="mailto:dubrowin.list@gmail.com">dubrowin.list@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div>Matt,
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<div>If your phone is using SIP, then you should enable sip inspection (7.x code or above) or fixup sip (6.x code) and have a rule that allows source (wherever you need) inbound on the outside interface to TCP 5060 (SIP port). The sip inspection or fixup should enable the proper ports for the require RTP streams. I had this working through an ASA at some point, but I don't remember if both ends were doing NAT or only one end. I don't know the phone you are talking about, but you also might want to look into STUN or ICE to get beyond the NAT Traversal issue, if that is what's causing the problem.
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<div>In the Firewall log, are you seeing Denys? or drops? Have you tried debug sip on the firewall console? I've been dealing with several ASA SIP issues lately. SIP trunking with NAT will certainly not work and there is a Cisco Bug that my company discovered when setting up our PBX.
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<div> Shlomo in Israel<br><br> </div>
<div><div class="Ih2E3d"><span class="gmail_quote">On 11/27/07, <b class="gmail_sendername">Matt</b> <<a href="mailto:mhoppes@gmail.com" target="_blank">mhoppes@gmail.com</a>> wrote:</span>
</div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;"><div class="Ih2E3d">Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to
<span name="st">Asterisk</span> behind a <span name="st">PIX</span> firewall?<br><br>Ports 10000-20000 UDP are open on the <span name="st">PIX</span> and forwarding to the
<span name="st">Asterisk</span> server. The <span name="st">Asterisk</span> server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra 9133i.
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