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ango, <br>
I have been playing with connecting two * servers... I had to stop but
I do think I had it working... even with this link:<br>
<a class="moz-txt-link-freetext" href="http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers">http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers</a><br>
it wasn't as straight forward as I would have liked... I used a
register on one box and a conf entry on the other. then I reversed the
config for the other * box<br>
<br>
pbx82 = 10.10.15.82<br>
pbx15 = 10.10.15.15<br>
<tt><br>
<b>on pbx15</b><br>
<br>
sip.conf<br>
register => <a class="moz-txt-link-abbreviated" href="mailto:sip_pbx15:1234@10.10.15.82">sip_pbx15:1234@10.10.15.82</a><br>
[sip_to_pbx82]<br>
type=user<br>
username=sip_pbx15<br>
accountcode=sip_from_pbx15<br>
secret=1234<br>
context=sip_from_pbx15<br>
host=10.10.15.82<br>
disallow=all<br>
allow=ulaw<br>
allow=alaw<br>
allow=gsm<br>
<br>
extensions.conf<br>
[sip_pbx15_to_pbx82]<br>
; dial a pbx82 extension via SIP with 982XXX where XXX is the extension<br>
</tt><tt>exten =>
_982XXX,1,Dial(<a class="moz-txt-link-abbreviated" href="mailto:SIP/sip_pbx15:1234@10.10.15.82/${EXTEN:3">SIP/sip_pbx15:1234@10.10.15.82/${EXTEN:3</a>},20,r)<br>
</tt><tt>;exten => _982XXX,1,Dial(SIP/${EXTEN:3},20,r)<br>
exten => _982XXX,n,Playback(connection-failed)<br>
exten => _982XXX,n,Playback(vm-goodbye)<br>
exten => _982XXX,n,Congestion<br>
exten => _982XXX,n,Hangup</tt><br>
<br>
<tt><b>on pbx82</b><br>
<br>
extensions.conf<br>
[sip_from_pbx15]<br>
exten => _XXX,1,Wait(1)<br>
exten => _XXX,n,Answer()<br>
exten => _XXX,n,Dial(SIP/${EXTEN},20,,r)<br>
exten => _XXX,n,VoiceMailMain<br>
exten => _XXX,n,Hangup()</tt><br>
<br>
[sip_from_pbx15] must be accessible in your inbound or default
context...<br>
I don't think I made any general section changes...<br>
<br>
it has been a few weeks since I played with it and I went only one
way... but if it worked one way it should work the other way too by
reverse duplicating the above config on pbx82 and pbx15 respectively.<br>
let me know how you make out...<br>
daveC<br>
<br>
<br>
Rilawich Ango wrote:
<blockquote
cite="mid6fbb529e0704200147u31e1f170u71985d823e8310d6@mail.gmail.com"
type="cite">I use realtime. Both information and extensions are
stored in DB. It
<br>
is just a simple setting of the user with dial plan "Dial(9003@S2)".
<br>
exten => 9003,1,Dial(9003@S2)
<br>
What I found is the following.
<br>
<br>
9002 ---> S1 ---> S2
<br>
9002 can make request to S1 and S1 forward the request to S2.
<br>
9002 ---> S1 <--- S2
<br>
S2 returns the mentioned error message to S1. (What I guess is 9002
<br>
only registers in S1 not in S2, so mentioned error message issued by
<br>
S2).
<br>
<br>
It is what I got from the above case. Do you have such configuration?
<br>
I have no idea to solve the problem
<br>
<br>
On 4/20/07, dave cantera <a class="moz-txt-link-rfc2396E" href="mailto:david.cantera@iacnet.net"><david.cantera@iacnet.net></a> wrote:
<br>
<blockquote type="cite">ango,
<br>
can you provide some sip.conf and extens.conf info?
<br>
daveC
<br>
<br>
Rilawich Ango wrote:
<br>
> hi,
<br>
> I have 2 asterisks with the following configuration.
<br>
> asterisk server 1 (S1) has an user 9002
<br>
> asterisk server 2 (S2) has an user 9003
<br>
> Both users can make call to each other without problem.
<br>
> Now I add both users to both servers, i.e.
<br>
> asterisk server 1 (S1) has users 9002,9003
<br>
> asterisk server 2 (S2) has users 9002,9003
<br>
> When 9002 dials 9003, Dial(SIP/9003@S2) or visa versa. Both
processes
<br>
> failed to make call with the following error.
<br>
> Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802
handle_response_invite:
<br>
> Failed to authenticate on INVITE to '"9002"
<br>
> <a class="moz-txt-link-rfc2396E" href="sip:90002@10.0.0.22"><sip:90002@10.0.0.22></a>;tag=as2ff0c493'
<br>
> Any solution to let them call each others?
<br>
> ango
<br>
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