<div> SIP requires RTP connections in addition to the signaling connection which normally happens on UDP 5060. The RTP connections vary in port usage (the range is configurable through rtp.conf) and are nearly impossible to get going without some "man in the middle" help when you have two Asterisk servers that are both behind NAT firewalls.
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<div> </div>
<div> If that's the case, you're much better off here with IAX where your signaling and media stream can be consolidated into a single stream. <br><br> </div>
<div><span class="gmail_quote">On 10/17/05, <b class="gmail_sendername">Michael Furdyk</b> <<a href="mailto:mfurdyk@takingitglobal.org">mfurdyk@takingitglobal.org</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div dir="ltr" align="left"><span><font face="Arial" color="#0000ff" size="2">Okay so it seems like it was the firewall, someone just suggested that we disable it (On Redhat server) and it's working fine... so does anyone know clearly what ports (other than 5060) SIP uses for these calls?
</font></span></div>
<div dir="ltr" align="left"><span><font face="Arial" color="#0000ff" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font face="Arial" color="#0000ff" size="2">-- Mike</font></span></div><br>
<div lang="en-us" dir="ltr" align="left">
<hr>
<font face="Tahoma" size="2"><b>From:</b> <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:
<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Michael Furdyk<br><b>Sent:</b>
October 17, 2005 2:54 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> [Asterisk-Users] SIP to SIP sadness<br></font><br> </div>
<div><span class="e" id="q_10700015705cf3ff_1">
<div></div>
<div dir="ltr" align="left"><span lang="EN-GB" style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><span>Wow, after getting the O'Reilly book delivered last week along with two Digium TDM400P's, I'm really getting the hang of this. But the SIP to SIP issue is still a problem... and it seems silly because everything else (should have been?) so much harder but is working pretty flawlessly. Basically I get no audio either way, and it tries to do a "native bridge" (handoff?)
</span></span></div>
<div dir="ltr" align="left"><span lang="EN-GB" style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><span></span></span> </div>
<div dir="ltr" align="left"><span lang="EN-GB" style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><span>So when I dial another SIP extension, I get:</span></span></div>
<div dir="ltr" align="left"><span lang="EN-GB" style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><span></span></span> </div>
<div dir="ltr" align="left"><span lang="EN-GB" style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><span></span></span> <font face="Arial" size="2">---<br> -- SIP/324-ab4d answered SIP/322-7e8d<br>We're at <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.1.195/" target="_blank">
192.168.1.195</a> port 16874<br>Answering with preferred capability 0x2 (gsm)<br>Answering with preferred capability 0x4 (ulaw)<br>Answering with non-codec capability 0x1 (telephone-event)<br>Reliably Transmitting (NAT) to
<a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.1.24:5060/" target="_blank">192.168.1.24:5060</a>:<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.1.24:5060/" target="_blank">
192.168.1.24:5060</a>;branch=z9hG4bK8E845F95F34044ACA77C54EF28288C32;received=<a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.1.24/" target="_blank">192.168.1.24</a>;rport=5060<br>From: Michael Furdyk <
<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip:322@192.168.1.195" target="_blank">sip:322@192.168.1.195</a>>;tag=411158625<br>To: <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip:324@192.168.1.195" target="_blank">
sip:324@192.168.1.195</a>>;tag=as6606adb1<br>Call-ID: <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24" target="_blank">28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24
</a><br>CSeq: 30931 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY<br>Contact: <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip:324@192.168.1.195" target="_blank">
sip:324@192.168.1.195</a>><br>Content-Type: application/sdp<br>Content-Length: 239</font></div>
<div><font face="Arial" size="2"></font> </div>
<div dir="ltr" align="left"><font face="Arial" size="2">v=0<br>o=root 3348 3348 IN IP4 <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.1.195/" target="_blank">192.168.1.195</a><br>s=session
<br>c=IN IP4 <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.1.195/" target="_blank">192.168.1.195</a><br>t=0 0<br>m=audio 16874 RTP/AVP 3 0 101<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:0 PCMU/8000
<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -</font></div>
<div><font face="Arial" size="2"></font> </div>
<div dir="ltr" align="left"><font face="Arial" size="2">---<br> -- Attempting native bridge of SIP/322-7e8d and SIP/324-ab4d</font></div>
<div><font face="Arial" size="2"></font> </div>
<div dir="ltr" align="left"><font face="Arial" size="2"><-- SIP read from <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.1.24:5060/" target="_blank">192.168.1.24:5060</a>: <br>ACK <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip:324@192.168.1.195" target="_blank">
sip:324@192.168.1.195</a> SIP/2.0<br>Via: SIP/2.0/UDP <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.1.24:5060/" target="_blank">192.168.1.24:5060</a>;rport;branch=z9hG4bKB644BADE71EF4422878597A96BE8D613
<br>From: Michael Furdyk <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip:322@192.168.1.195" target="_blank">sip:322@192.168.1.195</a>>;tag=411158625<br>To: <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip:324@192.168.1.195" target="_blank">
sip:324@192.168.1.195</a>>;tag=as6606adb1<br>Contact: <sip:322@192.168.1.24:5060><br>Call-ID: <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24" target="_blank">
28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24</a><br>CSeq: 30931 ACK<br>Max-Forwards: 70<br>Content-Length: 0</font></div>
<div dir="ltr" align="left"><font face="Arial" size="2"></font> </div>
<div dir="ltr" align="left"><span><font face="Arial" size="2">Here is my default in SIP.conf. Each SIP config has canreinvite=no</font></span></div>
<div dir="ltr" align="left"><span><font face="Arial" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font face="Arial" size="2">[general]<br>disallow=all<br>allow=gsm<br>allow=ulaw<br>nat=no<br>canreinvite=no<br>externip=(real external IP is here)<br>localnet=<a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.1.195/255.255.255.0" target="_blank">
192.168.1.195/255.255.255.0</a><br>srvlookup=yes<br>sipdebug=yes<br></font></span></div>
<div dir="ltr" align="left"><span><font face="Arial" size="2">I have tried nat=no and nat=yes</font></span></div>
<div dir="ltr" align="left"><span><font face="Arial" size="2"> </font></span></div></span></div><br>_______________________________________________<br>--Bandwidth and Colocation sponsored by <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://easynews.com/" target="_blank">
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