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<font face="monospace">Hi Wadii,<br>
<br>
The fact you are registering against a trunk provider does NOT
impact how you send the call to them (registration is mainly for
RECEIVING calls, and probably in this case a kind of auth method).
So, if you registered against provide.com SIP URI, you should send
the calls to the same SIP URI.<br>
<br>
Regards,<br>
</font>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com">https://www.opensips-solutions.com</a>
OpenSIPS Bootcamp 5-16 Dec 2022, online
<a class="moz-txt-link-freetext" href="https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/">https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/</a></pre>
<div class="moz-cite-prefix">On 11/20/22 1:42 PM, Wadii ELMAJDI |
Evenmedia wrote:<br>
</div>
<blockquote type="cite"
cite="mid:MR1P264MB3585E4494E4D8FDE8C1CAE99CE0B9@MR1P264MB3585.FRAP264.PROD.OUTLOOK.COM">
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<div style="font-family: Calibri, Arial, Helvetica, sans-serif;
font-size: 12pt; color: rgb(0, 0, 0); background-color: rgb(255,
255, 255);" class="elementToProof">
<span style="color: rgb(0, 0, 0); font-family: Calibri, Arial,
Helvetica, sans-serif;">I know i can route calls to any sip
uri , i'm just trying to find the most suitable solution.</span><br>
</div>
<div style="font-size: 12pt; background-color: rgb(255, 255,
255);" class="elementToProof">
<font face="Calibri, Arial, Helvetica, sans-serif"
color="#000000"><br>
</font></div>
<div style="font-size: 12pt; background-color: rgb(255, 255,
255);" class="elementToProof">
<font face="Calibri, Arial, Helvetica, sans-serif"
color="#000000">Its a usecase i never faced before,
<span style="background-color:rgb(255, 255,
255);display:inline !important" class="ContentPasted0">
on which the sip trunk provider requires a registration and
also challenging every outbound invite.<br>
</span><br>
Do you confirm it wouldnt be a bad idea running a registration
using uac_registrant module. which will allow me to receive
calls from the sip provider endpoint. At the same time add
their host/ip as a gateway in order to route outbound calls to
this same exact provider using DR module ?</font></div>
<div style="font-family:Calibri,Arial,Helvetica,sans-serif;
font-size:12pt; color:rgb(0,0,0)">
<br>
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<hr tabindex="-1" style="display:inline-block; width:98%">
<div id="divRplyFwdMsg" dir="ltr"><font style="font-size:11pt"
face="Calibri, sans-serif" color="#000000"><b>De :</b>
Bogdan-Andrei Iancu <a class="moz-txt-link-rfc2396E" href="mailto:bogdan@opensips.org"><bogdan@opensips.org></a><br>
<b>Envoyé :</b> vendredi 18 novembre 2022 11:27<br>
<b>À :</b> OpenSIPS users mailling list
<a class="moz-txt-link-rfc2396E" href="mailto:users@lists.opensips.org"><users@lists.opensips.org></a>; Wadii ELMAJDI | Evenmedia
<a class="moz-txt-link-rfc2396E" href="mailto:wadii@evenmedia.fr"><wadii@evenmedia.fr></a><br>
<b>Objet :</b> Re: [OpenSIPS-Users] relay invites UAC</font>
<div> </div>
</div>
<div><font face="monospace">Hi Wadii,<br>
<br>
You can route the call to any SIP URI you want :)<br>
<br>
Regards,<br>
</font>
<pre class="x_moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="https://www.opensips-solutions.com" data-auth="NotApplicable" class="x_moz-txt-link-freetext" moz-do-not-send="true">https://www.opensips-solutions.com</a>
OpenSIPS Bootcamp 5-16 Dec 2022, online
<a href="https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/" data-auth="NotApplicable" class="x_moz-txt-link-freetext" moz-do-not-send="true">https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/</a></pre>
<div class="x_moz-cite-prefix">On 11/15/22 10:14 PM, Wadii
ELMAJDI | Evenmedia wrote:<br>
</div>
<blockquote type="cite">
<div class="x_elementToProof"
style="font-family:Calibri,Arial,Helvetica,sans-serif;
font-size:12pt; color:rgb(0,0,0);
background-color:rgb(255,255,255)">
Hello Everyone,<br>
<br>
I am using opensips as sip proxy, receiving invites from my
clients (mostly b2bua), that i should relay to some
gateways.<br>
I am using Dynamic route module for such purpose.<br>
Some of those invites (depending on called DID or gateways
probing) should be relayed to a sip trunk service provider.<br>
My provider requires a registration, which i did using uac
registrant module.<br>
My question, is it possible to route calls to my sip
provider knowing it is neither a Gateway, nor a subscriber ?<br>
<br>
Thank you</div>
<br>
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</pre>
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