<div dir="ltr"><div dir="ltr">Hello,</div><div dir="ltr"><br></div><div dir="ltr">Could you share your Asterisk's sip context where you set up connectivity with telco? Just delete all usernames and passwords, if any....</div><div dir="ltr"><br></div><div dir="ltr">Your setup is very similar to the one I have with a very large telco in my country, and it works fine as long as you always keep Asterisk on the RTP path (no re-invites allowed, NAT always active)</div><div dir="ltr"><br><div><br clear="all"><div><div dir="ltr" class="gmail_signature"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><p style="color:rgb(0,0,0);margin:0px;font-family:Calibri,sans-serif;background-color:rgb(253,253,253)"><b><span style="font-family:Arial,sans-serif">Sérgio Charrua</span></b><br></p><div><div dir="ltr"><p style="font-size:11pt;color:rgb(0,0,0);margin:0px;font-family:Calibri,sans-serif;background-color:rgb(253,253,253)"><img src="http://cdn.shopify.com/s/files/1/0380/5305/t/11/assets/logo.png?11469850511917603945" width="96" height="32"><b><span style="font-size:9pt;font-family:Arial,sans-serif"><br></span></b></p><p style="font-size:11pt;color:rgb(0,0,0);margin:0px;font-family:Calibri,sans-serif;background-color:rgb(253,253,253)"><b><span style="font-size:9pt;font-family:Arial,sans-serif;color:red"><span style="color:rgb(0,0,0)"></span><span style="color:rgb(51,102,255)"><a href="http://www.voip.pt/" style="color:rgb(17,85,204)" target="_blank">www.voip.pt</a></span></span></b><span style="font-size:x-small"><span style="font-family:Arial,sans-serif;color:rgb(136,136,136)"><br></span><span style="font-family:Arial,sans-serif">Tel.: <span style="color:rgb(51,102,153)"><a href="callto:+351+91+104+12+66" style="color:rgb(51,102,153);text-decoration:none" target="_blank">+351 </a>21 130 71 77<br></span></span><span style="font-family:"Times New Roman",serif"></span></span></p><p style="font-size:11pt;color:rgb(0,0,0);margin:0px;font-family:Calibri,sans-serif;background-color:rgb(253,253,253)"><span style="font-size:x-small"><span style="font-family:Arial,sans-serif;color:rgb(187,187,187)"><span style="color:rgb(128,128,128)">Email : <strong><a href="mailto:sergio.charrua@voip.pt" style="color:rgb(17,85,204)" target="_blank">sergio.charrua@voip.pt</a></strong></span><br></span></span></p><table border="0" cellspacing="0" cellpadding="0" width="650" style="width:487.5pt">
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<span style="font-family:"Times New Roman",serif"></span></p><p class="MsoNormal"><span lang="EN-US"> </span></p><p style="color:rgb(0,0,0);margin:0px;font-family:Calibri,sans-serif;background-color:rgb(253,253,253)"> </p></div></div><div style="font-size:small"><br><br></div></div></div></div></div></div></div></div><br></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, May 11, 2021 at 8:02 AM David Villasmil <<a href="mailto:david.villasmil.work@gmail.com">david.villasmil.work@gmail.com</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"><div dir="ltr">So let me get this right:<div><br></div><div>asterisk (10.0.x.x)--->(192.168.0.192) proxy (10.0.x.x)--->(10.0.x.x)telco op</div><div><br></div><div>There's something i'm not seeing.</div><div>Can you explain further like i did above?</div><div><br></div><div><br clear="all"><div><div dir="ltr"><div dir="ltr"><div>Regards,</div><div><br></div>David Villasmil<div>email: <a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.com</a></div><div>phone: +34669448337</div></div></div></div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, May 10, 2021 at 8:42 PM Kashish Raheja <<a href="mailto:kashishraheja1809@gmail.com" target="_blank">kashishraheja1809@gmail.com</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"><div dir="ltr">Yes, the telecom operator is on the private network. 10.0.X.X is the SBC IP of the telecom operator to which we register. 10.0.X.X is reachable only through the second network interface. The complete flow is given below:<div><br></div><div><img src="cid:ii_koipy0tj0" alt="image.png" width="562" height="227"></div><div><br></div><div>Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first network interface IP, 10.0.87.230 is second network interface IP and 10.0.76.9 is telecom operator SBC IP to which we do SIP register.</div><div><br></div><div>We are running the rtpproxy on local IP (192.168.0.192) in the following way:</div><div><br></div><div><b>rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s udp:localhost:7722</b><br></div><div><br></div><div><br></div><div>Thanks.<br><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div>Regards</div><div>Kashish<br></div></div></div></div></div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <<a href="mailto:kashishraheja1809@gmail.com" target="_blank">kashishraheja1809@gmail.com</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"><div dir="ltr">Here are the SIP Traces:<div><br></div><div><b>Asterisk Server to Kamailio Server (SDP Packet):</b></div><div><br></div><div>2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP 3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060<br>Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944><br>Call-ID: <a href="mailto:58eb00885daef7ff3a67ad0e235e817a@14.98.22.110" target="_blank">58eb00885daef7ff3a67ad0e235e817a@14.98.22.110</a><br>From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944<br>To: <<a href="http://sip:09413745250@192.168.0.192:5060" target="_blank">sip:09413745250@192.168.0.192:5060</a>>;tag=aa2c806-Huku2c07186a1<br>CSeq: 102 INVITE<br>Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE<br>Contact: <sip:09413745250@10.0.X.X:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff><br>User-Agent: ZTE Softswitch/1.0.0<br>Require: timer<br>Session-Expires: 7200;refresher=uac<br>Content-Length: 182<br>Content-Type: application/sdp<br><br>v=0<br>o=- 1936 20890 IN IP4 10.0.X.X<br>s=SBC call<br>c=IN IP4 10.0.X.X<br>t=0 0<br>m=audio 37874 RTP/AVP 8 101<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=rtpmap:8 PCMA/8000/1<br></div><div><br></div><div><br></div><div><b>Kamailio Server to Telecom Operator Carrier (SDP Packet):</b></div><div><br></div><div>2021/05/10 15:54:52.835419 <a href="http://192.168.0.192:5060" target="_blank">192.168.0.192:5060</a> -> 3.X.X.X:5060<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060<br>Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944><br>Call-ID: <a href="mailto:58eb00885daef7ff3a67ad0e235e817a@14.98.22.110" target="_blank">58eb00885daef7ff3a67ad0e235e817a@14.98.22.110</a><br>From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944<br>To: <<a href="http://sip:09413745250@192.168.0.192:5060" target="_blank">sip:09413745250@192.168.0.192:5060</a>>;tag=aa2c806-Huku2c07186a1<br>CSeq: 102 INVITE<br>Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE<br>Contact: <sip:09413745250@10.0.X.X:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff><br>User-Agent: ZTE Softswitch/1.0.0<br>Require: timer<br>Session-Expires: 7200;refresher=uac<br>Content-Length: 182<br>Content-Type: application/sdp<br><br>v=0<br>o=- 1936 20890 IN IP4 10.0.X.X<br>s=SBC call<br>c=IN IP4 10.0.X.X<br>t=0 0<br>m=audio 37874 RTP/AVP 8 101<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=rtpmap:8 PCMA/8000/1<br></div><div><br></div><div><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div>Regards</div><div>Kashish</div></div></div></div></div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <<a href="mailto:kashishraheja1809@gmail.com" target="_blank">kashishraheja1809@gmail.com</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"><div dir="ltr">Hi All,<div><br></div><div>I have set up Kamailio in the following manner:</div><div><br></div><div>Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk) ---> Asterisk Server (on Cloud having public IP)</div><div><br></div><div>I am successfully able to route the call to Asterisk server on Cloud when I make a call to the number provided by the carrier and there is audio also on both sides.</div><div><br></div><div>However, when I am making an outbound call from Asterisk server to the number through Kamailio, there is no audio when I pick up the call. I have tried to capture the traces but not able to understand the exact problem here.</div><div><br></div><div>Note: I am running the RTP proxy on Kamailio server.</div><div><br></div><div>Any help on why this might be happening?</div><div><br></div><div>Thanks.<br clear="all"><div><div dir="ltr"><div dir="ltr"><div><div dir="ltr"><div>Regards</div><div>Kashish</div><div>+919413745250</div></div></div></div></div></div></div></div>
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