== Using UDPTL CoS mark 5 Entry point line 4875: peer is 9546020709. Entry point line 4875: peer is (null). == Using SIP RTP CoS mark 5 Entry point line 4875: peer is QMULTICOM. -- Executing [7862222222@from_nextone:1] NoOp("SIP/QMULTICOM-0000002f", "Call from Quintum to SVox") in new stack -- Executing [7862222222@from_nextone:2] Set("SIP/QMULTICOM-0000002f", "_SIP_CODEC_INBOUND=ulaw") in new stack -- Executing [7862222222@from_nextone:3] Set("SIP/QMULTICOM-0000002f", "_SIP_CODEC_OUTBOUND=g729") in new stack -- Executing [7862222222@from_nextone:4] Dial("SIP/QMULTICOM-0000002f", "SIP/SWITCHVOX/7862222222") in new stack == Using UDPTL CoS mark 5 Entry point line 4875: peer is SWITCHVOX. == Using SIP RTP CoS mark 5 [Sep 6 08:32:53] NOTICE[10153]: chan_sip.c:6274 try_suggested_sip_codec: Changing codec to 'g729' for this call because of ${SIP_CODEC} variable -- Called SIP/SWITCHVOX/7862222222 Entry point line 4875: peer is SWITCHVOX. -- SIP/SWITCHVOX-00000030 is ringing -- SIP/SWITCHVOX-00000030 answered SIP/QMULTICOM-0000002f [Sep 6 08:32:56] NOTICE[10153]: chan_sip.c:6274 try_suggested_sip_codec: Changing codec to 'ulaw' for this call because of ${SIP_CODEC} variable [Sep 6 08:32:56] NOTICE[10153]: chan_sip.c:6274 try_suggested_sip_codec: Changing codec to 'ulaw' for this call because of ${SIP_CODEC} variable Entry point line 4875: peer is SWITCHVOX. -- Locally bridging SIP/QMULTICOM-0000002f and SIP/SWITCHVOX-00000030 Entry point line 4875: peer is QMULTICOM. -- Executing [h@from_nextone:1] NoOp("SIP/QMULTICOM-0000002f", "") in new stack -- Executing [h@from_nextone:2] NoOp("SIP/QMULTICOM-0000002f", "RTPAUDIOQOS=ssrc=825398469;themssrc=9683;lp=0;rxjitter=0.000175;rxcount=1006;txjitter=0.000000;txcount=3003;rlp=0;rtt=0.010000") in new stack -- Executing [h@from_nextone:3] NoOp("SIP/QMULTICOM-0000002f", "RTPAUDIOQOSBRIDGED=ssrc=1041796395;themssrc=1959465842;lp=0;rxjitter=0.000302;rxcount=3003;txjitter=0.000000;txcount=333;rlp=0;rtt=65.536000") in new stack == Spawn extension (from_nextone, 7862222222, 4) exited non-zero on 'SIP/QMULTICOM-0000002f' Entry point line 4875: peer is SWITCHVOX. Entry point line 4875: peer is QMULTICOM. asterisk1-8*CLI> themssrc=9683 --> 0x25d3 SSRC from QUINTUM (204.9.236.18), SOURCE IP for this call ssrc=825398469 --> 0x313294c5 SSRC from ASTERISK (204.9.239.116) ON A-LEG Wireshark shows 1013 Received packets, asterisk reports 1006 and no loss. But there are 5 RTP packets that the Quintum sent after both legs of the call had been shutdown completely. Wireshark shows 3003(ulaw) + 2(g729) = 3005 Transmitted packets, asterisk reports 3003 and no loss. ssrc=1041796395 --> 0X3E188D2B SSRC from ASTERISK (204.9.239.116) ON B-LEG themssrc=195946584 --> 0x74cb1372 SSRC from SWITCHVOX (204.9.239.200), DESTINATION IP for this call Wireshark shows 3005 Received packets, asterisk reports 3003 and no loss Wireshark shows 333 Transmitted packets, asterisk reports 333 and no loss. Wireshark reports no packet loss for both legs and 55 RTCP Packets in total Wireshark reports that the QUINTUM (204.9.236.18) sent 10 RTCP Sender Reports and 9 Receiver Reports Wireshark reports that the ASTERISK (204.9.239.116) sent 12 RTCP Sender Reports and 0 Receiver Reports (A-leg) Quintum: Sender report 1 at 10.229413 seconds into the capture, packet 186, reporting 51 packets sent, none lost. Up to this point the latest seq# is 50, the first was 0, so indeed there were 51 packets sent Sender report 1 at 13.458555 seconds into the capture, packet 650, reporting 157 packets sent, none lost. Up to this point the latest seq# is 156, the first was 0, so indeed there were 157 packets sent Wireshark reports that the ASTERISK (204.9.239.116) sent 6 RTCP Sender Reports and 6 Receiver Reports (B-leg) Wireshark reports that the SWITCHVOX (204.9.239.200) sent 12 RTCP Sender Reports and 0 Receiver Reports (A-leg) Switchvox: