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<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>Look in rtp.conf. You
must have the same udp-ports open as the settings in rtp.conf<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>Anders<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>
<hr size=2 width="100%" align=center tabindex=-1>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span lang=EN-US
style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=2 face=Tahoma><span lang=EN-US style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Michael Furdyk<br>
<b><span style='font-weight:bold'>Sent:</span></b> den 17 oktober 2005 21:02<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: [Asterisk-Users] SIP
to SIP sadness</span></font><span lang=EN-US><o:p></o:p></span></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=blue face=Arial><span lang=DA
style='font-size:10.0pt;font-family:Arial;color:blue'>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?</span></font><span lang=DA><o:p></o:p></span></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=DA
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=blue face=Arial><span lang=DA
style='font-size:10.0pt;font-family:Arial;color:blue'>-- Mike</span></font><span
lang=DA><o:p></o:p></span></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=DA
style='font-size:12.0pt'><o:p> </o:p></span></font></p>
<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>
<hr size=2 width="100%" align=center tabIndex=-1>
</span></font></div>
<p class=MsoNormal style='margin-bottom:12.0pt'><b><font size=2 face=Tahoma><span
lang=EN-US style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=2 face=Tahoma><span lang=EN-US style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Michael Furdyk<br>
<b><span style='font-weight:bold'>Sent:</span></b> October 17, 2005 2:54 PM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users] SIP to
SIP sadness</span></font><span lang=EN-US><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>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?)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=DA
style='font-size:12.0pt'> </span></font><font size=2 face=Arial><span
lang=EN-GB style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>So when I dial another SIP extension, I get:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=DA
style='font-size:12.0pt'> </span></font><font size=2 face=Arial><span
lang=EN-GB style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=DA
style='font-size:12.0pt'> </span></font><font size=2 face=Arial><span
lang=DA style='font-size:10.0pt;font-family:Arial'>---<br>
-- SIP/324-ab4d answered SIP/322-7e8d<br>
We're at 192.168.1.195 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 192.168.1.24:5060:<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 192.168.1.24:5060;branch=z9hG4bK8E845F95F34044ACA77C54EF28288C32;received=192.168.1.24;rport=5060<br>
From: Michael Furdyk <sip:322@192.168.1.195>;tag=411158625<br>
To: <sip:324@192.168.1.195>;tag=as6606adb1<br>
Call-ID: <a href="mailto:28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24">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: <sip:324@192.168.1.195><br>
Content-Type: application/sdp<br>
Content-Length: 239</span></font><font size=2 face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=DA
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
</div>
<p class=MsoNormal><font size=2 face=Arial><span lang=DA style='font-size:10.0pt;
font-family:Arial'>v=0<br>
o=root 3348 3348 IN IP4 192.168.1.195<br>
s=session<br>
c=IN IP4 192.168.1.195<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 - - - -</span></font><span lang=DA><o:p></o:p></span></p>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=DA
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
</div>
<p class=MsoNormal><font size=2 face=Arial><span lang=DA style='font-size:10.0pt;
font-family:Arial'>---<br>
-- Attempting native bridge of SIP/322-7e8d and SIP/324-ab4d</span></font><span
lang=DA><o:p></o:p></span></p>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=DA
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
</div>
<p class=MsoNormal><font size=2 face=Arial><span lang=DA style='font-size:10.0pt;
font-family:Arial'><-- SIP read from 192.168.1.24:5060: <br>
ACK sip:324@192.168.1.195 SIP/2.0<br>
Via: SIP/2.0/UDP
192.168.1.24:5060;rport;branch=z9hG4bKB644BADE71EF4422878597A96BE8D613<br>
From: Michael Furdyk <sip:322@192.168.1.195>;tag=411158625<br>
To: <sip:324@192.168.1.195>;tag=as6606adb1<br>
Contact: <sip:322@192.168.1.24:5060><br>
Call-ID: <a href="mailto:28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24">28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24</a><br>
CSeq: 30931 ACK<br>
Max-Forwards: 70<br>
Content-Length: 0</span></font><span lang=DA><o:p></o:p></span></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=DA
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=DA style='font-size:10.0pt;
font-family:Arial'>Here is my default in SIP.conf. Each SIP config has
canreinvite=no</span></font><span lang=DA><o:p></o:p></span></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=DA
style='font-size:12.0pt'> <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=DA style='font-size:10.0pt;
font-family:Arial'>[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=192.168.1.195/255.255.255.0<br>
srvlookup=yes<br>
sipdebug=yes</span></font><span lang=DA><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=DA style='font-size:10.0pt;
font-family:Arial'>I have tried nat=no and nat=yes</span></font><span lang=DA><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=DA style='font-size:10.0pt;
font-family:Arial'> <o:p></o:p></span></font></p>
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