<div dir="ltr"><div><div><div>i know what is the exactly problem. i enable debug for h323 and it says: <br></div>"could not find user by name 200 or address 192.168.0.146"<br><br></div>when i change "peer-146" to "200" every thing is ok and i can call from two side. but it is not good for me because 200 is the name of extension and when i config asterisk systems, i don't know the name of extensions, therefore i should use addresses not name of extensions. <br>
</div>do you know how i should define address of the other end in h323.conf file? i define the address by "host=192.168.0.146" but asterisk can not find it? why? <br></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad <span dir="ltr"><<a href="mailto:asghar144@gmail.com" target="_blank">asghar144@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">please post cli output for both calls.</div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div class="h5">On Mon, Apr 22, 2013 at 11:32 AM, s m <span dir="ltr"><<a href="mailto:sam.gh1986@gmail.com" target="_blank">sam.gh1986@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5">hello everybody<br>
<br>
i want to have sip connection between two asterisk systems (145 and<br>
146). connection from 145 to 146 is ok but i can not call from 146 to<br>
145.<br>
this is h323.conf file in 145:<br>
[peer146]<br>
host=192.168.0.146<br>
type=friend<br>
context=from-trunk<br>
<br>
<br>
[to-146]<br>
type=peer<br>
host=192.168.0.146<br>
faststart=yes<br>
tunneling=no<br>
progress_audio=yes<br>
disallow=all<br>
allow=alaw<br>
allow=ulaw<br>
<br>
this is mu extensions.conf file in 145:<br>
<br>
[from-trunk]<br>
exten=>_1.,1,Dial(SIP/to-231/1${EXTEN:1})<br>
[line-231]<br>
exten=>_2.,1,Dial(H323/to-146/2${EXTEN:1})<br>
<br>
i have this error: dropping call because extensions '100', 's' and 'i'<br>
doesn't exists in context default".<br>
<br>
if i change "peer146" to "general", every thing is ok and i can call<br>
from two side. my question is: in h323 connection, is it a MUST to<br>
have "general" context in h323.conf? if not, why i have this error and<br>
how i can solve it?<br>
thanks in advance<br>
sam<br>
<br></div></div><span class="HOEnZb"><font color="#888888">
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</font></span></blockquote></div><br></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br></div>