<div dir="ltr">Kashish,<div><br></div><div>Best you can do is take a trace on BOTH ends and share it with us.</div><div><br clear="all"><div><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><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 Thu, May 27, 2021 at 2:26 PM Kashish Raheja <<a href="mailto:kashishraheja1809@gmail.com">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"><div>Haven't been able to sort this out yet. Anything am I missing here?</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></div></div></div></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, May 21, 2021 at 1:44 AM 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"><div dir="ltr">Hi Daniel,<div><br></div><div>Sorry it took some time for me to make these changes.</div><div><br></div><div>I have made all the changes as suggested by you however it still doesn't seem to work. No audio in the outbound call however incoming call works fine.</div><div><br></div><div>Here are the SIP traces after making the changes:</div><div><br></div><div><b>INVITE: Asterisk to Kamailio:</b></div><div><pre style="color:rgb(51,51,51);margin-top:0px;margin-bottom:0px;line-height:16.25px"> │INVITE <a href="http://sip:09413745250@192.168.0.192:5060" target="_blank">sip:09413745250@192.168.0.192:5060</a> SIP/2.0
<a href="http://3.236.72.101:5060" target="_blank">3.236.72.101:5060</a> <a href="http://192.168.0.192:5060" target="_blank">192.168.0.192:5060</a> <a href="http://10.0.87.230:5060" target="_blank">10.0.87.230:5060</a> <a href="http://10.0.76.9:5060" target="_blank">10.0.76.9:5060</a> │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;rport
──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Max-Forwards: 70
01:22:15.782149 │ <b>INVITE (SDP) </b> │ │ │ │From: <<a href="http://sip:68983619@192.168.0.192:5060" target="_blank">sip:68983619@192.168.0.192:5060</a>>;tag=as69eb1cce
+0.050579 │ <b>──────────────────────────></b> │ │ │ │To: <<a href="http://sip:09413745250@192.168.0.192:5060" target="_blank">sip:09413745250@192.168.0.192:5060</a>>
01:22:15.832728 │ 100 trying -- your call is │ │ │ │Contact: <<a href="http://sip:68983619@3.236.72.101:5060" target="_blank">sip:68983619@3.236.72.101:5060</a>>
+0.000348 │ <────────────────────────── │ │ │ │Call-ID: <a href="mailto:1191aedf331ec3e35955bf376a20999d@14.98.22.110" target="_blank">1191aedf331ec3e35955bf376a20999d@14.98.22.110</a>
01:22:15.833076 │ │ │ INVITE (SDP) │ │CSeq: 102 INVITE
+0.004863 │ │ │ ──────────────────────────> │ │User-Agent: Asterisk PBX 17.7.0
01:22:15.837939 │ │ │ 100 Trying │ │Date: Thu, 20 May 2021 19:52:15 GMT
+0.799120 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Supported: replaces, timer
+0.000179 │ │ │ <────────────────────────── │ │P-Preferred-Identity: <<a href="mailto:sip%3A68983600@10.0.76.9" target="_blank">sip:68983600@10.0.76.9</a>>
01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp
+1.490537 │ <────────────────────────── │ │ │ │Content-Length: 263
01:22:18.127775 │ │ │ 180 Ringing │ │
+0.000189 │ │ │ <────────────────────────── │ │v=0
01:22:18.127964 │ 180 Ringing │ │ │ │o=root 1560151942 1560151942 IN IP4 3.236.72.101 <b>(Asterisk's Public IP)</b>
+0.349351 │ <────────────────────────── │ │ │ │s=Asterisk PBX 17.7.0
01:22:18.477315 │ │ │ 180 Ringing │ │c=IN IP4 3.236.72.101 <b>(Asterisk's Public IP)</b>
+0.000206 │ │ │ <<<──────────────────────── │ │t=0 0
01:22:18.477521 │ 180 Ringing │ │ │ │m=audio 14046 RTP/AVP 8 0 101
+19.181387 │ <<<──────────────────────── │ │ │ │a=rtpmap:8 PCMA/8000
01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:0 PCMU/8000
+0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:101 telephone-event/8000
01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=fmtp:101 0-16
+0.241852 │ <────────────────────────── │ │ │ │a=maxptime:150
01:22:37.974479 │ ACK │ │ │ │a=sendrecv
+0.000282 │ ──────────────────────────> │ │ │ │
01:22:37.974761 │ │ │ ACK │ │
+4.095171 │ │ │ ──────────────────────────> │ │
01:22:42.069932 │ │ │ BYE │ │
+0.000361 │ │ │ <────────────────────────── │ │
01:22:42.070293 │ BYE │ │ │ │
+0.244125 │ <────────────────────────── │ │ │ │
01:22:42.314418 │ 200 OK │ │ │ │
+0.000275 │ ──────────────────────────> │ │ │ │
01:22:42.314693 │ │ │ 200 OK │ │
│ │ │ ──────────────────────────> │ │</pre></div><div><br></div><div><br></div><div><b>INVITE: Kamailio to Telco:</b></div><div><pre style="color:rgb(51,51,51);margin-top:0px;margin-bottom:0px;line-height:16.25px"> │INVITE <a href="mailto:sip%3A09413745250@10.0.76.9" target="_blank">sip:09413745250@10.0.76.9</a> SIP/2.0
<a href="http://3.236.72.101:5060" target="_blank">3.236.72.101:5060</a> <a href="http://192.168.0.192:5060" target="_blank">192.168.0.192:5060</a> <a href="http://10.0.87.230:5060" target="_blank">10.0.87.230:5060</a> <a href="http://10.0.76.9:5060" target="_blank">10.0.76.9:5060</a> │Record-Route: <sip:192.168.0.192;lr=on;ftag=as69eb1cce>
──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP 192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.0
01:22:15.782149 │ INVITE (SDP) │ │ │ │Via: SIP/2.0/UDP 3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060
+0.050579 │ ──────────────────────────> │ │ │ │Max-Forwards: 69
01:22:15.832728 │ 100 trying -- your call is │ │ │ │From: <<a href="http://sip:68983619@192.168.0.192:5060" target="_blank">sip:68983619@192.168.0.192:5060</a>>;tag=as69eb1cce
+0.000348 │ <────────────────────────── │ │ │ │To: <<a href="http://sip:09413745250@192.168.0.192:5060" target="_blank">sip:09413745250@192.168.0.192:5060</a>>
01:22:15.833076 │ │ │ <b> INVITE (SDP) </b> │ │Contact: <<a href="http://sip:68983619@3.236.72.101:5060" target="_blank">sip:68983619@3.236.72.101:5060</a>>
+0.004863 │ │ │ <b>──────────────────────────></b> │ │Call-ID: <a href="mailto:1191aedf331ec3e35955bf376a20999d@14.98.22.110" target="_blank">1191aedf331ec3e35955bf376a20999d@14.98.22.110</a>
01:22:15.837939 │ │ │ 100 Trying │ │CSeq: 102 INVITE
+0.799120 │ │ │ <────────────────────────── │ │User-Agent: Asterisk PBX 17.7.0
01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Date: Thu, 20 May 2021 19:52:15 GMT
+0.000179 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Supported: replaces, timer
+1.490537 │ <────────────────────────── │ │ │ │P-Preferred-Identity: <<a href="mailto:sip%3A68983600@10.0.76.9" target="_blank">sip:68983600@10.0.76.9</a>>
01:22:18.127775 │ │ │ 180 Ringing │ │Content-Type: application/sdp
+0.000189 │ │ │ <────────────────────────── │ │Content-Length: 279
01:22:18.127964 │ 180 Ringing │ │ │ │
+0.349351 │ <────────────────────────── │ │ │ │v=0
01:22:18.477315 │ │ │ 180 Ringing │ │o=root 1560151942 1560151942 IN IP4 10.0.87.230 <b>(RTP Proxy's private IP)</b>
+0.000206 │ │ │ <<<──────────────────────── │ │s=Asterisk PBX 17.7.0
01:22:18.477521 │ 180 Ringing │ │ │ │c=IN IP4 10.0.87.230 <b>(RTP Proxy's private IP)</b>
+19.181387 │ <<<──────────────────────── │ │ │ │t=0 0
01:22:37.658908 │ │ │ 200 OK (SDP) │ │m=audio 37322 RTP/AVP 8 0 101
+0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:8 PCMA/8000
01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=rtpmap:0 PCMU/8000
+0.241852 │ <────────────────────────── │ │ │ │a=rtpmap:101 telephone-event/8000
01:22:37.974479 │ ACK │ │ │ │a=fmtp:101 0-16
+0.000282 │ ──────────────────────────> │ │ │ │a=maxptime:150
01:22:37.974761 │ │ │ ACK │ │a=sendrecv
+4.095171 │ │ │ ──────────────────────────> │ │a=nortpproxy:yes
01:22:42.069932 │ │ │ BYE │ │
+0.000361 │ │ │ <────────────────────────── │ │
01:22:42.070293 │ BYE │ │ │ │
+0.244125 │ <────────────────────────── │ │ │ │
01:22:42.314418 │ 200 OK │ │ │ │
+0.000275 │ ──────────────────────────> │ │ │ │
01:22:42.314693 │ │ │ 200 OK │ │
│ │ │ ──────────────────────────> │ │</pre></div><div><br></div><div><br></div><div><b>On 200: Kamailio to Asterisk:</b></div><div><pre style="color:rgb(51,51,51);margin-top:0px;margin-bottom:0px;line-height:16.25px"> │SIP/2.0 200 OK
<a href="http://3.236.72.101:5060" target="_blank">3.236.72.101:5060</a> <a href="http://192.168.0.192:5060" target="_blank">192.168.0.192:5060</a> <a href="http://10.0.87.230:5060" target="_blank">10.0.87.230:5060</a> <a href="http://10.0.76.9:5060" target="_blank">10.0.76.9:5060</a> │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060
──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Record-Route: <sip:192.168.0.192;lr;ftag=as69eb1cce>
01:22:15.782149 │ INVITE (SDP) │ │ │ │Call-ID: <a href="mailto:1191aedf331ec3e35955bf376a20999d@14.98.22.110" target="_blank">1191aedf331ec3e35955bf376a20999d@14.98.22.110</a>
+0.050579 │ ──────────────────────────> │ │ │ │From: <<a href="http://sip:68983619@192.168.0.192:5060" target="_blank">sip:68983619@192.168.0.192:5060</a>>;tag=as69eb1cce
01:22:15.832728 │ 100 trying -- your call is │ │ │ │To: <<a href="http://sip:09413745250@192.168.0.192:5060" target="_blank">sip:09413745250@192.168.0.192:5060</a>>;tag=aa2c806-t7ln3f58c3ea1
+0.000348 │ <────────────────────────── │ │ │ │CSeq: 102 INVITE
01:22:15.833076 │ │ │ INVITE (SDP) │ │Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
+0.004863 │ │ │ ──────────────────────────> │ │Contact: <sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
01:22:15.837939 │ │ │ 100 Trying │ │User-Agent: ZTE Softswitch/1.0.0
+0.799120 │ │ │ <────────────────────────── │ │Require: timer
01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Session-Expires: 7200;refresher=uac
+0.000179 │ │ │ <────────────────────────── │ │Content-Length: 208
01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp
+1.490537 │ <────────────────────────── │ │ │ │
01:22:18.127775 │ │ │ 180 Ringing │ │v=0
+0.000189 │ │ │ <────────────────────────── │ │o=- 1026 13186 IN IP4 192.168.0.192 <b>(RTP Proxy's public IP)</b>
01:22:18.127964 │ 180 Ringing │ │ │ │s=SBC call
+0.349351 │ <────────────────────────── │ │ │ │c=IN IP4 192.168.0.192 <b>(RTP Proxy's public IP)</b>
01:22:18.477315 │ │ │ 180 Ringing │ │t=0 0
+0.000206 │ │ │ <<<──────────────────────── │ │m=audio 48462 RTP/AVP 8 101
01:22:18.477521 │ 180 Ringing │ │ │ │a=rtpmap:101 telephone-event/8000
+19.181387 │ <<<──────────────────────── │ │ │ │a=fmtp:101 0-15
01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:8 PCMA/8000/1
+0.073719 │ │ │ <────────────────────────── │ │a=nortpproxy:yes
01:22:37.732627 │ <b> 200 OK (SDP)</b> │ │ │ │
+0.241852 │ <b><──────────────────────────</b> │ │ │ │
01:22:37.974479 │ ACK │ │ │ │
+0.000282 │ ──────────────────────────> │ │ │ │
01:22:37.974761 │ │ │ ACK │ │
+4.095171 │ │ │ ──────────────────────────> │ │
01:22:42.069932 │ │ │ BYE │ │
+0.000361 │ │ │ <────────────────────────── │ │
01:22:42.070293 │ BYE │ │ │ │
+0.244125 │ <────────────────────────── │ │ │ │
01:22:42.314418 │ 200 OK │ │ │ │
+0.000275 │ ──────────────────────────> │ │ │ │
01:22:42.314693 │ │ │ 200 OK │ │
│ │ │ ──────────────────────────> │ │</pre></div><div><br></div><div><br></div><div>On the cloud Asterisk, all the relevant public IPs are already allowed.</div><div><br></div><div>Have run the rtpproxy on the bridge mode with the following command:</div><div><b>/usr/local/bin/rtpproxy -s udp:<a href="http://127.0.0.1:7722" target="_blank">127.0.0.1:7722</a> -u asterisk -p /var/run/rtpproxy/rtpproxy.pid -l <a href="http://192.168.0.192/10.0.87.230" target="_blank">192.168.0.192/10.0.87.230</a></b><br></div><div><br></div><div>Apart from this, in the Asterisk console I can see that the RTP packets are being sent to Kamailio</div><div><br></div><div><pre style="color:rgb(51,51,51);margin-top:0px;margin-bottom:0px;line-height:16.25px">Sent RTP packet to <a href="http://14.98.22.110:11648" target="_blank">14.98.22.110:11648</a> (type 08, seq 022310, ts 029280, len 000160)
Sent RTP packet to <a href="http://14.98.22.110:11648" target="_blank">14.98.22.110:11648</a> (type 08, seq 022311, ts 029440, len 000160)
Sent RTP packet to <a href="http://14.98.22.110:11648" target="_blank">14.98.22.110:11648</a> (type 08, seq 022312, ts 029600, len 000160)
Sent RTP packet to <a href="http://14.98.22.110:11648" target="_blank">14.98.22.110:11648</a> (type 08, seq 022313, ts 029760, len 000160)</pre></div><div><br></div><div>However, there isn't any log for receiving the RTP packets unlike for incoming calls</div><div><br></div><div>Anything am I missing here?</div><div><br></div><div>Thanks.<br clear="all"><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div>Regards</div><div>Kashish</div></div></div></div></div></div></div><div class="gmail_quote"><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"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
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