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Page 176 of Asterisk, the definitive manual, discusses "Connecting
an Asterisk system to a SIP provider" in the context of, at least
the concept of, "trunking".<br>
<br>
Previously, I wasn't able to connect to the peer, and attributed
that to a combination of double NAT (plus), and latency and lag due
to wi-fi. Now that I'm directly connected to the cable modem (well,
gateway router and modem combo), the connection is better and I'm
able to make outgoing VoIP calls with Jitsi.<br>
<br>
Am I right in thinking that the very same connection parameters I
entered in Jitsi will work fine when entered in Asterisk with syntax
like:<br>
<br>
register => <a class="moz-txt-link-abbreviated" href="mailto:username:password@your.provider.tld">username:password@your.provider.tld</a><br>
<br>
and by creating the peer entry in sip.conf for the service provider.<br>
<br>
One concern is that the provider uses:<br>
<br>
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<ol>
<li> User ID can be any one of your 11-digit babyTEL telephone
numbers. Typically your main number but can be any one of your
other phone numbers. </li>
<li> For your protection the SIP Password field does not reveal
your password until you explicitly click on ‘Show password’. </li>
<li>If Outbound Proxy is not supported on your system, try one of
the following two options:
<ol style="list-style:lower-alpha">
<li>Add the line “198.38.7.34 sip.babytel.ca” to your system’s
“hosts” file and configure the SIP Proxy as:
“sip.babytel.ca:5065”. This uses the TCP/IP “hosts” file
address mapping mechanism to redirect SIP traffic to the
Outbound Proxy. </li>
<li> Configure the SIP Proxy as: “198.38.7.34:5065”. This
replaces the SIP Proxy address with a resolved Outbound
Proxy address. </li>
</ol>
</li>
</ol>
<p><br>
On a mac, I added that line to the hosts file -- but I'm not sure
it's required. How do I specify the SIP proxy and the outbound
proxy? What's the distinction between a SIP proxy and outbound
proxy?<br>
</p>
<p><br>
<br>
</p>
In Jitsi, I configured as <a class="moz-txt-link-abbreviated" href="mailto:123456789@sip.babytel.ca">123456789@sip.babytel.ca</a> for SIP id.<br>
<br>
In "Connection" I used "sip.babytel.ca" for the registrar and the
user, 1234567890, as the the authorization name. I put the proxy as
nat5.babytel.ca, port 5065 and the preferred transport as UDP. I
don't see all those options, particularly surrounding the proxy and
outbound proxy. Again, I'm unclear on why there's a proxy
specificed, and then a different outbound proxy is specified as
well.<br>
<br>
<br>
<br>
<br>
How do I establish that I've entered the parameters correctly in
Asterisk? Or, how do I establish that the parameters are
incorrectly entered? Because Jitsi is able to call out and in, I
believe that eliminates NAT, firewall or other networking issues. <br>
<br>
<br>
<br>
thanks,<br>
<br>
Thufir<br>
<br>
<br>
<br>
<br>
<br>
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