<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 TRANSITIONAL//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; CHARSET=UTF-8">
<META NAME="GENERATOR" CONTENT="GtkHTML/3.24.5">
</HEAD>
<BODY>
SIP-registration errors are solved by restarting the Asterisk-server. But I expect them to return in time... <BR>
<BR>
I can make call now, but the other end does not hear me. So problem with RTP-flow...<BR>
<BR>
Can someone guide me to the solution ?<BR>
<BR>
On the Asterisk-server I have this (iptables):<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">-A RH-Firewall-1-INPUT -p udp --dport 4569 -j ACCEPT</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">-A RH-Firewall-1-INPUT -p tcp --dport 5060 -j ACCEPT</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">-A RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">-A RH-Firewall-1-INPUT -p udp --dport 11000:11500 -j ACCEPT</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">-A RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 25 -j ACCEPT</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j ACCEPT</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">-A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited</FONT></FONT><BR>
<BR>
In rtp.conf I have this :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">rtpstart=11000</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">rtpend=11500</FONT></FONT><BR>
<BR>
Asterisk is behind firewall. Endian firewall has following configuration :<BR>
<BR>
enable SIP proxy transparant<BR>
RTP port low : 11000<BR>
RTP port high : 11500<BR>
<BR>
Firewall port forwarding : uplink:5060 >>> asterisk_ip:5060<BR>
<BR>
Asterisk himself says :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff"> -- Executing [050510484@intern:1] NoOp("SIP/grandstream-09813b58", "via 3StarsNet") in new stack</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff"> -- Executing [050510484@intern:2] Dial("SIP/grandstream-09813b58", "SIP/3starsnet/050510484") in new stack</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff"> -- Called 3starsnet/050510484</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff"> -- SIP/3starsnet-0981bf08 is making progress passing it to SIP/grandstream-09813b58</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff"> -- SIP/3starsnet-0981bf08 answered SIP/grandstream-09813b58</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff"> == Spawn extension (intern, 050510484, 2) exited non-zero on 'SIP/grandstream-09813b58'</FONT></FONT><BR>
<BR>
What do I need in sip.conf to overcome these rtp-problems ??<BR>
I have :<BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">externip=78.21.62.99</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">canreinvite=no</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">jbenable = yes</FONT></FONT><BR>
<BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[3starsnet]</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">type=peer</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">...</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">nat=yes</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">...</FONT></FONT><BR>
<BR>
<BR>
Thanks for the help !<BR>
<BR>
Jonas.<BR>
<BR>
<BR>
On Thu, 2009-06-25 at 17:25 +0200, jonas kellens wrote:<BR>
<BLOCKQUOTE TYPE=CITE>
Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2)<BR>
<BR>
Can someone see why SIP-registration fails ??
</BLOCKQUOTE>
</BODY>
</HTML>