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Yo,<br>
<br>
can you monitor the traffic between and attach it to us (for xample
with wireshark)? maybe it would be clearer then if it's nat issue, or
something else?<br>
<br>
J.<br>
<br>
Jonas Gauffin wrote:
<blockquote
cite="mid:d021d3be0801080620w10ceb937jfabafcdf1b027f52@mail.gmail.com"
type="cite">
<pre wrap="">Back to basics. I've reverted to the default sip profile configs.
I've now specified "192.168.1.102" as the domain (the FS ip). Both
phones do now register using that ip.
The registration goes OK.
But when I call from one phone to another (both on the same lan as
FS), I only get audio one way.
I call 1203 which brigdes to sofia/192.168.1.102/conny in the dialplan.
This must be the simplest test case, right?
On Jan 8, 2008 2:00 PM, Brian West <a class="moz-txt-link-rfc2396E" href="mailto:brian.west@mac.com"><brian.west@mac.com></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Btw if you put back the config to the default. It will work because
the ip is the domain and the packet don't loop thur your nat router
needlessly.
/b
Sent from my iPhone
On Jan 8, 2008, at 7:39 AM, Jonas Gauffin <a class="moz-txt-link-rfc2396E" href="mailto:jonas.gauffin@gmail.com"><jonas.gauffin@gmail.com></a>
wrote:
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">* I only get audio one way when dialing from one registered UA to
another, check attached log internal_call.log
</pre>
</blockquote>
<pre wrap="">Chances are mydomain.com is your public IP right? and I suspect NAT
is
involved? If so then you need to setup a profile for Natted phones
behind the same nat. One profile for phones outside the nat to
register with FreeSWITCH ... and the public one listed below.
</pre>
</blockquote>
<pre wrap="">mydomain.com points on my public ip, correct. The phones and FS are
behind a router (on the same lan).
FS have loaded the three profiles (nat, default and outbound). I have
made no changes to them.
Currently I have just tested with phones that is on the same lan as
FS.
here is a registration trans:
REGISTER sip:mydomain.com SIP/2.0
Via: SIP/2.0/UDP
192.168.1.123:
5060;rport;branch=z9hG4bK6f95bcc8e01f69dda9f5ebf92adc2c93
From: <a class="moz-txt-link-rfc2396E" href="mailto:sip:tobbe@mydomain.com"><sip:tobbe@mydomain.com></a>;tag=66055002
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:tobbe@mydomain.com"><sip:tobbe@mydomain.com></a>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:58ee65fb0b696f5a370d839058387f2c@192.168.1.123">58ee65fb0b696f5a370d839058387f2c@192.168.1.123</a>
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:tobbe@192.168.1.123:5060"><sip:tobbe@192.168.1.123:5060></a>
CSeq: 53 REGISTER
Expires: 300
Authorization: Digest username="tobbe",realm="mydomain.com",[...
some info stripped out
...],uri="sip:mydomain.com",nc=00000001,algorithm=MD5
User-Agent: CMI CM5K
Content-Length: 0
tobbe has user_context set to "default" in conf/directory/default/
tobbe.xml
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">
* Inbound external calls do not work, i get [407][Proxy
Authentication Required], check log external_call.log
</pre>
</blockquote>
<pre wrap="">This is why you need to start another profile to accept calls from
the
outside world with authentication turned off. That would fix that
problem.
</pre>
</blockquote>
<pre wrap="">well. the nat profile exists and it has auth_calls set to false. but
maybe it isnt used in my config?
</pre>
<blockquote type="cite">
<pre wrap="">I honestly like having multiple UA's it solves many problems.
</pre>
</blockquote>
<pre wrap="">what do u mean by that? I'm using multiple UA's, all of them are
currently on the same LAN as FS right now (testing enviroment), but
later on, they may also be elsewhere.
But currently I'm trying to get two UA's to be able to talk with each
other (as I said, the audio is just going one way).
And I'm not saying that it's one UA to rule them all.. =) I'm just
trying to understand this all new configuration how a profile is
selected (for inbound/outbound external calls, internal calls,
internal registration (on the same lan as FS) and external
registration (not on the same lan as FS)).
thanks,
jonas
</pre>
</blockquote>
<blockquote type="cite">
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<pre wrap=""><!---->
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</pre>
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