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<DIV><SPAN class=204244019-29122005><FONT face=Arial color=#0000ff size=2>The
word from Kevin Fleming and Digium is that the use of realtime to support
multiple Asterisk boxes sharing sip is not supported or even known to work at
this point.</FONT></SPAN></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B> Asterisk
[mailto:Asterisk@isgcom.com]<BR><B>Sent:</B> Thursday, December 29, 2005 12:14
PM<BR><B>To:</B> asterisk-users@lists.digium.com<BR><B>Subject:</B>
[Asterisk-Users] Realtime Multiple Asterisk boxes andrtcachefriends
MWI<BR><BR></FONT></DIV>
<DIV><FONT face=Arial color=#000000 size=2>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt">I am
working an a multiple box asterisk solution.<SPAN
style="mso-spacerun: yes"> </SPAN>I need phones to be able to
login to multiple asterisk servers.<SPAN style="mso-spacerun: yes">
</SPAN>I need Phone A to be able to register to switch A and call Phone B that
is registered to switch B. <o:p></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><o:p> </o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt">With
rtcachfriends=no<SPAN style="mso-spacerun: yes"> </SPAN>this can be
done, However I then loss MWI and “sip show peers” plus if a Phone becomes
unreachable the phone I get dead air until the dial timeout reached.
<o:p></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><o:p> </o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt">With
rtcachfriends=yes<SPAN style="mso-spacerun: yes"> </SPAN>I get MWI &
“Sip show peers”, However I cannot call phones that register to a different
switch. <SPAN
style="mso-spacerun: yes"> </SPAN><o:p></o:p></SPAN></P><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><o:p>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"></SPAN> </P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt">My
current working solution is to have rtcachfriends=yes. Place the call
via sip if dialstatus= chanunavaliable </SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt">I then
route the call to the other switch via an IAX trunk. Everything
works but I don't have a true load balance soltuion. Plus it really only works
for 2 boxes. It get out of hand when I add more.. </SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"></o:p></SPAN><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"></SPAN> </P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt">I have
tried using AGI and dialing the “full contact” found in the SIP realtime
table. It works if the phone is active, but if the phone is no active I get
dead air until the dial timeout is reached. This will not work as I cannot
have 12 sec of dead air.<SPAN style="mso-spacerun: yes"> </SPAN><SPAN
style="mso-spacerun: yes"> </SPAN>So is there a way know the status
of a SIP UA?<SPAN style="mso-spacerun: yes"> </SPAN>It is it in the SIP
realtime data? <SPAN style="mso-spacerun: yes"> </SPAN>I looked at
regseconds but it does not seem to be it because I can have a UA that is
unreachable and the regseconds are not expired.<SPAN
style="mso-spacerun: yes"> </SPAN></SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><o:p></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"></SPAN> </P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt">Could
realtime be altered to add a status filed to the SIP realtime
table? <SPAN style="mso-spacerun: yes"> </SPAN></SPAN><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><o:p> </o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"></SPAN><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><o:p></o:p></SPAN> </P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt">Or is
there a asterisk configuration option that I missed?<SPAN
style="mso-spacerun: yes">
</SPAN></SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><SPAN
style="mso-spacerun: yes"></SPAN></SPAN> </P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><SPAN
style="mso-spacerun: yes">This is my first post so please forgive me if I
posted this in the wrong list. </SPAN></SPAN><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><o:p> </o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><o:p></o:p></SPAN> </P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt"><o:p></o:p></SPAN> </P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt">Many
thanks!<o:p></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0in 0in 0pt"><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-bidi-font-size: 12.0pt">Doug
Gillespie<o:p></o:p></SPAN></P></FONT></DIV></BLOCKQUOTE></BODY></HTML>