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<DIV dir=ltr align=left><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p><SPAN
class=224024903-17102005>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></o:p></SPAN></DIV>
<DIV dir=ltr align=left><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p><SPAN
class=224024903-17102005></SPAN></o:p></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p><SPAN
class=224024903-17102005>So when I dial another SIP extension, I
get:</SPAN></o:p></SPAN></DIV>
<DIV dir=ltr align=left><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p><SPAN
class=224024903-17102005></SPAN></o:p></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p><SPAN
class=224024903-17102005></SPAN></o:p></SPAN> <FONT face=Arial
size=2>---<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</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
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 - - -
-</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
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</FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr align=left><SPAN class=224024903-17102005><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 class=224024903-17102005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=224024903-17102005><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=192.168.1.195/255.255.255.0<BR>srvlookup=yes<BR>sipdebug=yes<BR></FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=224024903-17102005><FONT face=Arial size=2>I
have tried nat=no and nat=yes</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=224024903-17102005><FONT face=Arial
size=2> </DIV></FONT></SPAN></BODY></HTML>