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<p>Hi Joshua<br>
</p>
<div class="moz-cite-prefix">Le 08/04/2020 à 15:28, Joshua C. Colp a
écrit :<br>
</div>
<blockquote type="cite"
cite="mid:CAM0A2Z0pDx3TQj7s6X_GjCCMDPz7XsFDMo3H6VnySgwjrGHh+A@mail.gmail.com">
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<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Mon, Apr 6, 2020 at 2:06
PM Administrator <<a href="mailto:admin@tootai.net"
moz-do-not-send="true">admin@tootai.net</a>> wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left:1px solid
rgb(204,204,204);padding-left:1ex">Hello,<br>
<br>
We have a provider which is using Kamailio as front end. Our
asterisk <br>
13/chan_sip server has no problem to register and
pass/receive calls <br>
form this provider.<br>
<br>
Now we want to move to asterisk 16/pjsip and face problem.
Registration <br>
is OK but when we pass a call our INVITE never receive
answer from the <br>
provider. We opened a ticket to their support but in the
mean time we <br>
want to know if someone is using successfully a PJSIP
channel against <br>
Kamailio.<br>
<br>
Another one: despite the fact that they use 5061 port, it's
not TLS but <br>
UDP. Our asterisk16 has no TLS configured.<br>
<br>
We use wizard which looks like:<br>
<br>
[Provider-tootai](!)<br>
;<br>
type = wizard<br>
sends_auth = yes<br>
sends_registrations = yes<br>
accepts_auth = no<br>
accepts_registrations = no<br>
endpoint/call_group = 1<br>
endpoint/pickup_group = 1<br>
endpoint/accountcode = TOOTAi<br>
endpoint/language = fr<br>
endpoint/allow = !all,ulaw,alaw,g729<br>
endpoint/context = incoming-Provider<br>
endpoint/direct_media = no<br>
endpoint/dtmf_mode = inband<br>
registration/retry_interval = 20<br>
registration/max_retries = 0<br>
registration/expiration = 3600<br>
registration/transport = transport-udp<br>
aor/max_contacts = 2<br>
aor/qualify_frequency = 2000<br>
<br>
[Provider](Provider-tootai)<br>
;<br>
remote_hosts = <a href="http://sips.provider.eu"
rel="noreferrer" target="_blank" moz-do-not-send="true">sips.provider.eu</a><br>
endpoint/callerid = "TOOTAi" <00xx xxx xxx xxx><br>
aor/contact = sip:<a href="http://sips.provider.eu:5061"
rel="noreferrer" target="_blank" moz-do-not-send="true">sips.provider.eu:5061</a><br>
registration/client_uri = <a
href="mailto:sips%3AOUR_ID@sips.provider.eu"
target="_blank" moz-do-not-send="true">sips:OUR_ID@sips.provider.eu</a><br>
registration/server_uri = sips:<a
href="http://sips.provider.eu:5061" rel="noreferrer"
target="_blank" moz-do-not-send="true">sips.provider.eu:5061</a><br>
outbound_auth/username = OUR_ID<br>
outbound_auth/password = OUR_PWD<br>
identity/match = PROVIDER_IP<br>
</blockquote>
<div><br>
</div>
<div>Your server URI For registration and calling differs in
that one uses "sips" and the other "sip" for URI scheme. Is
there a particular reason they differ? I'd also expect
"sips" not to be used at all if it's strictly UDP. You could
also compare chan_sip and chan_pjsip traffic to see what the
difference is.<br>
</div>
</div>
</div>
</blockquote>
<p>Yes, someone point this error and I correct it. As said in my
previous message, I had to add outbound_proxy to make it work in
UDP. Provideer support gave me false information by saying that
port 5061 was for UDP but it was as usually for TLS. I correct all
the stuff, had to modify openssl.cnf and downgrade it to TLSv1 as
they still use this one and now connection is OK in UDP as well as
TLS.<br>
</p>
<p>Thanks for your support<br>
</p>
<pre class="moz-signature" cols="72">--
Daniel</pre>
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