29 Sep
2005
29 Sep
'05
2:59 p.m.
Matt,
You must have user=peer (I think friend will also work) in the [ser] section
of the sip.conf in order for Asterisk to find and dial that 4005 extension
in your [proxy] context.
That exten should then setup a SIP call to 4005(a)192.168.0.10 for an incoming
RURI of 4005@xxxxxxxxxx. Assuming that IP is SER then SER will need to do a
lookup for that user and domain to find the contact address. The outgoing
RURI to SER for the 2 leg will be 4005(a)192.168.0.10. Your SER setup need to
deal with routing the calls via Asterisk in this way.
A general. complete solution with SER as registrar and proxy and Asterisk as
B2BUA Feature Server for SIP PBX is beyond the scope of this email :). I've
done this for a client under NDA so I cant just publish all of their ser and
Asterisk configs. I dont own them. Perhaps there are other on this list that
are free to publish that sort of very detailed configuration info.
I've found it very difficult to find specific help for difficult problems on
mailing lists. Most of the hard questions go unanswered, likely due to NDAs
and conflicts of interest. A consultant that puts food on the table by
supporting open-source software is unlikely to give his 'product" away for
free.
The SER community seems much better about this and the core developers
answer a lot of hard questions on the various mailing lists.
The "core" developers for Asterisk are Digium and they sell support for a
living so dont expect too much free support. Getting the software for free
is a pretty good bargin , IMO, even if there is no 'free' support.
Mark
On 9/29/05, Matt L. Zhu <coder0000(a)hotmail.com> wrote:
in my dialplan, i have this
[proxy] # same as the context in sip.conf
exten => 4005.,1,Dial(SIP/${EXTEN}(a)192.168.0.10)
i am new to asterisk, how can i make it so the exten will route the call
to
the other sipphone connected to the ser proxy.
i really want to achieve sipphone->ser->asterisk->ser->sipphone when a
phone
calls another. just getting confused how exten will reroute to ser again.
<BLOCKQUOTE style='PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT:
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From: <i>Mark Aiken <aiken.mark(a)gmail.com></i><br>Reply-To<br>Reply-To:
<i>Mark
Aiken <aiken.mark(a)gmail.com></i><br>To/i><br>To: <i>Iqbal
<iqbal(a)gigo.co.uk></i><br>CC/i><br>CC: <i>Bogdan-Andrei Iancu
<bogdan(a)voice-system.ro>ro>, "Matt L. Zhu"
<coder0000(a)hotmail.com>om>, users(a)openser.org</i><br>Subjectr>Subject:
<i>Re:
[Users] Re: [Devel] openser and asterisk</i><br>Date: <i>Thu, 29 Sep
2005
12:04:21 -0500</i><br>
<br>You may want to set type=peer in the [ser] section. Also , I assume
you
have a Dial statement in your 'proxy' context in the dialplan. You need
that to connect the 2 users. We have no problems using Asterisk as a
sip server with ser or openser as the registrar and proxy. I think
there are many using this kind of setup so it does work.<br>
<br>
Mark<br><br><div><span class="gmail_quote">On 9/29/05,
<b
class="gmail_sendername">Iqbal</b> <<a
href="mailto:iqbal@gigo.co.uk">iqbal@gigo.co.uk</a>>
wrote:</span><blockquote class="gmail_quote" style="margin:0pt
0pt 0pt
0.8ex;padding-left:1ex">
whats is sip debug on asterisk showing<br><br>Bogdan-Andrei Iancu
wrote:<br><br>> Hi Matt,<br>><br>> I redirected this
email on the
users mailing list - it's more<br>> appropriate.<br>><br>>
the idea
seams ok, with couple of comments:
<br>> 1) be sure that fwd to localhost is ok (instead of a routable
IP)<br>> 2) doing Record-Route may be a good think.<br>><br>>
to
debug tour problem, add some log("...") statements into your
script
<br>> to be able to trace the processing. Also a network trace (including
on<br>> lo device) will be helpful to see what happens - if the messages
are<br>> received, if they are sent and where. Also watch the log for
potential
<br>> errors.<br>><br>> regards,<br>>
bogdan<br>><br>><br>><br>> Matt L. Zhu
wrote:<br>><br>>> has anyone successfully setup openser as the
frontend proxy for<br>>> asterisk? here is my setup
<br>>><br>>> /etc/asterisk/sip.conf<br>>>
[general]<br>>> context=default<br>>>
port=5065<br>>>
bindaddr=<a href="http://0.0.0.0">0.0.0.0
<http://0.0.0.0></a><br>>>
srvlookup=yes<br>>><br>>> [ser]
<br>>> type=user<br>>> context=proxy<br>>>
host=<a
href="http://192.168.0.10">192.168.0.10
<http://192.168.0.10></a><br>>><br>>>
then i
edited openser.cfg to do something like this<br>>><br>>>
if
<br>>>
(uri=~"sip:[a-zA-Z\.]*@(xxx\.xxx\.com)|(192\.168\.0\.10)")
{<br>>> forward(
localhost, 5065 );<br>>> break;<br>>>
};<br>>><br>>> i connected two sipphones (wengo) in this
case to openser, but calls<br>>> are not going through at all,
connecting directly to asterisk works.
<br>>> have anyone worked in this
situation?<br>>><br>>>
thanks<br>>><br>>><br>>><br>>>
_______________________________________________<br>>> Devel mailing
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