Yair Hakak
2006-Feb-05 05:55 UTC
[Asterisk-Users] re: questions about sip requests to asterisk 1.2
hi all,
I keep asking the question and getting no replies, so i'll keep asking :-)
In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request
from SER, specifically
rewritehostport("myIP:5070"); (asterisk running on port 5070)
asterisk
picks up the request and matches it to the dialplan, i.e. if in ser i was
sending to 151@myServer, it will make it 151@myIP:5070, and asterisk will
match it to 151 in the dialplan.
In asterisk 1.2 asterisk completely ignores the request (even at most
verbose level) and an ngrep shows a "not found" returned to SER.
anyone have any idea why this is happening, bug/feature, or how to get it to
work the way it did in 1.09? I want to upgrade but I don't want to lose
this functionality.
thanks for any help,
yair
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Jean-Michel Hiver
2006-Feb-05 06:03 UTC
[Asterisk-Users] re: questions about sip requests to asterisk 1.2
> In asterisk 1.2 asterisk completely ignores the request (even at most > verbose level) and an ngrep shows a "not found" returned to SER. > > anyone have any idea why this is happening, bug/feature, or how to get > it to work the way it did in 1.09? I want to upgrade but I don't want > to lose this functionality.Since I use 1.0.9 and use exactly the same scheme, I am interested on how to upgrade as well. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
C. Zerbo
2006-Feb-05 12:47 UTC
[Asterisk-Users] re: questions about sip requests to asterisk 1.2
<html><div style='background-color:'><P>you need to
setup a asterisk peer at port 5070 in sip.conf to get the callreplying correctly
to ser.<BR><BR></P><BR><BR><BR>
<DIV></DIV>
<DIV>Cheick Zerbo</DIV>
<DIV>Corbimas.com</DIV>
<DIV><A
href="mailto:cz@corbimas.com">cz@corbimas.com</A></DIV>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT:
#a0c6e5 2px solid; MARGIN-RIGHT: 0px"><FONT style="FONT-SIZE:
11px; FONT-FAMILY: tahoma,sans-serif">
<HR color=#a0c6e5 SIZE=1>
From: <I>Yair Hakak <yair@hakak.com></I><BR>Reply-To:
<I>Asterisk Users Mailing List - Non-Commercial
Discussion<asterisk-users@lists.digium.com></I><BR>To:
<I>Asterisk Users List
<asterisk-users@lists.digium.com></I><BR>Subject:
<I>[Asterisk-Users] re: questions about sip requests to asterisk
1.2</I><BR>Date: <I>Sun, 5 Feb 2006 14:55:32
+0200</I><BR><BR>
<DIV>hi all,</DIV>
<DIV> I keep asking the question and getting no replies, so
i'll keep asking :-)</DIV>
<DIV> </DIV>
<DIV> In asterisk 1.09, with autocreatepeer=yes, if i send
asterisk a SIP request from SER, specifically </DIV>
<DIV> rewritehostport("myIP:5070"); (asterisk running
on port 5070) asterisk picks up the request and matches it to the dialplan, i.e.
if in ser i was sending to <A
href="mailto:151@myServer">151@myServer</A>, it will make it
<A href="mailto:151@myIP:5070">151@myIP:5070</A>, and
asterisk will match it to 151 in the dialplan. </DIV>
<DIV> </DIV>
<DIV>In asterisk 1.2 asterisk completely ignores the request (even at most
verbose level) and an ngrep shows a "not found" returned
to SER.</DIV>
<DIV> </DIV>
<DIV>anyone have any idea why this is happening, bug/feature, or how to
get it to work the way it did in 1.09? I want to upgrade but I
don't want to lose this functionality.</DIV>
<DIV> </DIV>
<DIV>thanks for any help,</DIV>
<DIV> yair <BR> </DIV>
<DIV> </DIV><BR>
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