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<p>Hello,<br>
</p>
<br>
<div class="moz-cite-prefix">On 12.10.17 12:30, 赵国杰 wrote:<br>
</div>
<blockquote type="cite"
cite="mid:6dc1a3e1.de22.15f10229170.Coremail.zhaoguojie2010@163.com">
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style="line-height:1.7;color:#000000;font-size:14px;font-family:Arial">
<div>Hello,</div>
<div> In a standard sip flow, the call goes like: sip user A
--> kamailio --> pstn --> landline user B. However,
when user A has a bad internet access, the audio is broken. So
what I want is to let sip user A send a invite to kamailio
first, then kamailio send invite to user A and B's landline
number through pstn, then bridge the two call together. </div>
<div> I understand this can be achieved by using FREESWITCH
originate and bridge command. I've tried but there's no audio
both ways, which really makes me feel stupid of myself. So I'm
wondering if this can be done with kamailio? If so, how?</div>
</div>
</blockquote>
kamailio cannot bridge a call like freeswitch does, because kamailio
doesn't handle media stream.<br>
<br>
The only option that you can do in kamailio to connect two phones is
to rely on REFER for doing a call transfer. See dialog module and
dlg bridge feature from there, maybe it is enough for what you want
to achieve.<br>
<br>
Cheers,<br>
Daniel<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="moz-txt-link-abbreviated" href="http://www.twitter.com/miconda">www.twitter.com/miconda</a> -- <a class="moz-txt-link-abbreviated" href="http://www.linkedin.com/in/miconda">www.linkedin.com/in/miconda</a>
Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - <a class="moz-txt-link-abbreviated" href="http://www.asipto.com">www.asipto.com</a>
Kamailio World Conference - <a class="moz-txt-link-abbreviated" href="http://www.kamailioworld.com">www.kamailioworld.com</a></pre>
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