Re: [SR-Users] No Audio For Outbound Calls

Hi Daniel, Sorry it took some time for me to make these changes. 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. Here are the SIP traces after making the changes: *INVITE: Asterisk to Kamailio:* │INVITE sip:09413745250@192.168.0.192:5060 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;rport ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Max-Forwards: 70 01:22:15.782149 │ *INVITE (SDP) * │ │ │ │From: <sip:68983619@192.168.0.192:5060>;tag=as69eb1cce +0.050579 │ *──────────────────────────>* │ │ │ │To: <sip:09413745250@192.168.0.192:5060> 01:22:15.832728 │ 100 trying -- your call is │ │ │ │Contact: <sip:68983619@3.236.72.101:5060> +0.000348 │ <────────────────────────── │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d(a)14.98.22.110 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: <sip:68983600@10.0.76.9> 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 *(Asterisk's Public IP)* +0.349351 │ <────────────────────────── │ │ │ │s=Asterisk PBX 17.7.0 01:22:18.477315 │ │ │ 180 Ringing │ │c=IN IP4 3.236.72.101 *(Asterisk's Public IP)* +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 │ │ │ │ │ ──────────────────────────> │ │ *INVITE: Kamailio to Telco:* │INVITE sip:09413745250@10.0.76.9 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │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: <sip:68983619@192.168.0.192:5060>;tag=as69eb1cce +0.000348 │ <────────────────────────── │ │ │ │To: <sip:09413745250@192.168.0.192:5060> 01:22:15.833076 │ │ │ * INVITE (SDP) * │ │Contact: <sip:68983619@3.236.72.101:5060> +0.004863 │ │ │ *──────────────────────────>* │ │Call-ID: 1191aedf331ec3e35955bf376a20999d(a)14.98.22.110 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: <sip:68983600@10.0.76.9> 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 *(RTP Proxy's private IP)* +0.000206 │ │ │ <<<──────────────────────── │ │s=Asterisk PBX 17.7.0 01:22:18.477521 │ 180 Ringing │ │ │ │c=IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +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 │ │ │ │ │ ──────────────────────────> │ │ *On 200: Kamailio to Asterisk:* │SIP/2.0 200 OK 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │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: 1191aedf331ec3e35955bf376a20999d(a)14.98.22.110 +0.050579 │ ──────────────────────────> │ │ │ │From: <sip:68983619@192.168.0.192:5060>;tag=as69eb1cce 01:22:15.832728 │ 100 trying -- your call is │ │ │ │To: <sip:09413745250@192.168.0.192:5060>;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 *(RTP Proxy's public IP)* 01:22:18.127964 │ 180 Ringing │ │ │ │s=SBC call +0.349351 │ <────────────────────────── │ │ │ │c=IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 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 │ * 200 OK (SDP)* │ │ │ │ +0.241852 │ *<──────────────────────────* │ │ │ │ 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 │ │ │ │ │ ──────────────────────────> │ │ On the cloud Asterisk, all the relevant public IPs are already allowed. Have run the rtpproxy on the bridge mode with the following command: */usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 <http://127.0.0.1:7722> -u asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230 <http://192.168.0.192/10.0.87.230>* Apart from this, in the Asterisk console I can see that the RTP packets are being sent to Kamailio Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts 029280, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts 029440, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts 029600, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts 029760, len 000160) However, there isn't any log for receiving the RTP packets unlike for incoming calls Anything am I missing here? Thanks. Regards Kashish >