how to check if my server is hack?
Posted: Tue Nov 19, 2019 12:07 am
Hi all,
Ealier today I receive a call from one of our VoIP Providers saying that we've made around $1000(AU/US not sure) worth of international calls, how can we check if we really made this call coming from our server? First off this provider is under talk and our company have not committed yet to their service because we can't make outgoing calls, all the time when making an outgoing calls this is the message appearing in asteriks cli:
"[Oct 10 16:13:06] -- Executing [976861386580543@default:2] Dial("SIP/999-00137154", "SIP/aatroxcommunications/61386580543") in new stack
[Oct 10 16:13:06] == Using SIP RTP CoS mark 5
[Oct 10 16:13:06] WARNING[13398][C-0047e583]: chan_sip.c:6276 sip_call: No audio format found to offer. Cancelling call to 61386580543
[Oct 10 16:13:06] -- Couldn't call SIP/aatroxcommunications/61386580543
[Oct 10 16:13:06] == Everyone is busy/congested at this time (0:0/0/0)"
Please badly need help in reviewing this, $1000 surely is a lot of money.
Ealier today I receive a call from one of our VoIP Providers saying that we've made around $1000(AU/US not sure) worth of international calls, how can we check if we really made this call coming from our server? First off this provider is under talk and our company have not committed yet to their service because we can't make outgoing calls, all the time when making an outgoing calls this is the message appearing in asteriks cli:
"[Oct 10 16:13:06] -- Executing [976861386580543@default:2] Dial("SIP/999-00137154", "SIP/aatroxcommunications/61386580543") in new stack
[Oct 10 16:13:06] == Using SIP RTP CoS mark 5
[Oct 10 16:13:06] WARNING[13398][C-0047e583]: chan_sip.c:6276 sip_call: No audio format found to offer. Cancelling call to 61386580543
[Oct 10 16:13:06] -- Couldn't call SIP/aatroxcommunications/61386580543
[Oct 10 16:13:06] == Everyone is busy/congested at this time (0:0/0/0)"
Please badly need help in reviewing this, $1000 surely is a lot of money.