<div dir="ltr">I know I promised it... but... I FIGURED IT OUT!!!<div><br></div><div>So, I'll just reply so that the next one with this issue can find this</div><div><br></div><div>The only bit of information was using fromuser instead of defaultuser... that only thing did the trick... so</div>
<div><br></div><div><div>[vpsserver]</div><div>type=friend</div><div>host=<a href="http://vpsserver.domain.com">vpsserver.domain.com</a></div><div>context=inbound</div><div>fromuser=myuser</div><div>secret=mypass</div></div>
<div><br></div><div>on the client's sip.conf did the trick</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Apr 18, 2014 at 9:24 AM, Gervasio Marchand Cassataro <span dir="ltr"><<a href="mailto:gmc@gmc.uy" target="_blank">gmc@gmc.uy</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">I'm not exactly nailing it in participation on my thread, so I swear this is my last message if no one replies ;)<div>
<br></div><div>I went ahead and enabled full debugging, and got some interesting results (available at <a href="http://pastebin.com/KiY6DMHi" target="_blank">http://pastebin.com/KiY6DMHi</a>)</div>
<div><br></div><div>What I see is:</div><div><ol><li>The registration works fine</li><li>When the client tries to establish a call to the server, the server looks into either the From or Contact headers... but the client sends the caller id in that place and that's when the INVITE gets rejected</li>
</ol><div>I guess a proper question would be "Is there any way on the sip.conf to specify the contact (I think that s@ip would work, as that's registered on the client) or on the pjsip.conf to whitelist the ip of a registered contact? something like insecure=invite"</div>
</div><div><br></div><div>Thanks!</div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Apr 16, 2014 at 8:02 PM, Gervasio Marchand Cassataro <span dir="ltr"><<a href="mailto:gmc@gmc.uy" target="_blank">gmc@gmc.uy</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">Just a heads up... Enabled NOTICEs on the server and I see this every 10 seconds or so<div><div>
<pre><code>[Apr 16 18:58:28] NOTICE[2138]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '"asterisk" <<a href="mailto:sip%3Aasterisk@179.25.158.95" target="_blank">sip:asterisk@179.25.158.95</a>>' failed for '<a href="http://179.25.158.95:5060" target="_blank">179.25.158.95:5060</a>' (callid: <a href="http://477ca2fd0db3a5542dcf2afd50673b89@179.25.158.95:5060" target="_blank">477ca2fd0db3a5542dcf2afd50673b89@179.25.158.95:5060</a>) - No matching endpoint found</code></pre>
Thanks in advance for any help / ideas / clues or something! I'm scratching my head around this and at this point</div></div></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Wed, Apr 16, 2014 at 6:26 PM, Gervasio Marchand Cassataro <span dir="ltr"><<a href="mailto:gmc@gmc.uy" target="_blank">gmc@gmc.uy</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">It's my first post here, so I'll cut to the chase<div><br></div><div><p>I have 2 Asterisk servers and want to connect them using sip on one
and pjsip on the other one. One is running at home and another at a VPS.
The first one will be the client (with dynamic ip) and the 2nd the
server.</p>
<p>The client uses sip and the server pjsip.</p>
<p>This is the client's sip.conf</p>
<pre><code>[general]
context = default
allowguest = no
realm = <a href="http://myrealm.com" target="_blank">myrealm.com</a>
udpbindaddr = 0.0.0.0
qualify = yes
subscribecontext = default
localnet=<a href="http://192.168.1.0/255.255.255.0externhost=myhost.com" target="_blank">192.168.1.0/255.255.255.0
externhost=myhost.com</a>
externrefresh=30
dtmfmode = auto
canreinvite = no
jbenable = no
sendrpid = yes
trustrpid = no
disallow=all
allow=ulaw
allow=alaw
register => myuser:mypass@vpsserver
[vpsserver]
type=friend
secret=myuser
defaultuser=mypass
host=<a href="http://vpsserver.domain.com" target="_blank">vpsserver.domain.com</a>
context=inbound
canreinvite=no
insecure=port,invite
</code></pre>
<p>And this is the server's pjsip.conf</p>
<pre><code>[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
[home]
type=endpoint
context=trusted
disallow=all
allow=ulaw
allow=alaw
transport=transport-udp
auth=home
aors=home
[home]
type=auth
auth_type=userpass
password=mypass
username=myuser
[home]
type=aor
max_contacts=10
</code></pre>
<p>When I check on the client, executing <code>sip show registry</code> I get</p>
<pre><code>Host dnsmgr Username Refresh State Reg.Time
vpsserver:5060 N myuser 104 Registered Tue, 15 Apr 2014 22:57:34
</code></pre>
<p>which I guess means everything is ok... on the client side, I have on my extensions.conf</p>
<pre><code>exten => 66,1,Dial(SIP/1@vpsserver)
</code></pre>
<p>and on the server's extensions.conf (in the trusted context) I have</p>
<pre><code>exten => 1,1,Playback(hello-world)
</code></pre>
<p>So far so good... but when I dial 66 on my client Asterisk, I see the
following SIP dialog on the server... the only weird thing is that I
see some From: 192.168.1.112 (that's my home Asterisk's internal IP...
the externhost works fine for all the providers I'm using, so I doubt
that's an issue)</p><p><a href="http://pastebin.com/hkFezB8j" target="_blank">http://pastebin.com/hkFezB8j</a><br></p><p>Thanks in advance!</p></div></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>