Calls not coming in after Database Migration
Posted: Fri Oct 11, 2019 4:44 am
Hello,
Installation Method: Scratch Install on centos 6.9 final, method followed (Document provided by Ray Solomon. Asterisk version 11.22.0-vici). 1x Dedicated Server from Oneprovider.
Server Specs;
CPU 2x Intel Xeon E5-2620 - 2 GHz - 6 core(s)
RAM
128GB - DDR4
STORAGE
3x 120GB (SSD SATA)
BANDWIDTH
Unmetered @ 1Gbps
-------------------------------
Hello experts, I have recently migrated database from a running server to a new server. (Both server specs and installation method is same). I followed all the steps required to get things going on the new server as i have done this a lot of times previously without facing any major issues and everything seems to start smoothly after migration. But this time things are acting a bit wierd. Initially everything seems to be normal like i can login, i'm also able to register the carrier, get the softphone online and also able to dial manual calls via Eyebeam registered to a vicidial phone ext to thension. The weirdness starts when we login to the campaign I see calls connecting and going into the survey but then nothing happens, like I am not able to see a single call entering the survey or the call menu(IVR). The call flow is like number gets answered enters the 8373 survey where it plays sip-silence for 1 second and enters a call menu where it gets the treatment, hears the recording and follows the instructions. Now the point is when the call enters the 8373 survey i see it on the asterisk console but unable to see any calls in survey or IVR on Realtime screen. I tried Predictive Dialer extension 8369 and 8368 still i dont see any LIVE calls towards the available agent. below is a snippet from asterisk.
[Oct 11 02:27:06] -- Executing [8368@default:1] Playback("SIP/usa-0000021a", "sip-silence") in new stack
[Oct 11 02:27:06] -- Executing [h@default:1] AGI("Local/52317619256919696@default-0000025f;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----0") in new stack
[Oct 11 02:27:06] -- <SIP/usa-0000021a> Playing 'sip-silence.gsm' (language 'en')
[Oct 11 02:27:06] > 0x7f83ac016d40 -- Probation passed - setting RTP source address to 216.221.155.204:59578
[Oct 11 02:27:07] -- Executing [8368@default:2] AGI("SIP/usa-0000021a", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 11 02:27:07] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=EBP3))
[Oct 11 02:27:07] -- <SIP/usa-0000021a>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 11 02:27:07] -- Executing [8368@default:3] AGI("SIP/usa-0000021a", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Oct 11 02:27:07] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 11 02:27:07] -- <SIP/usa-0000021a>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 11 02:27:07] -- Executing [8368@default:4] AGI("SIP/usa-0000021a", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Oct 11 02:27:07] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 11 02:27:07] -- <SIP/usa-0000021a>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 11 02:27:07] -- Executing [8368@default:5] Hangup("SIP/usa-0000021a", "") in new stack
[Oct 11 02:27:07] == Spawn extension (default, 8368, 5) exited non-zero on 'SIP/usa-0000021a'
[Oct 11 02:27:07] -- Executing [h@default:1] AGI("SIP/usa-0000021a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Oct 11 02:27:07] -- <SIP/usa-0000021a>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 11 02:27:08] -- <Local/52317619256919696@default-0000025f;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --20-----0 completed, returning 0
can u please help me rectify this issue or identify what is making the calls hangup? above snippet is when i switched it to ext 8368 to make any available call land on the agent but nothing happened instead the voip minutes are being used like normal but i dont know where the calls are going or whats wrong. please get me a fix or advice for this as its really urgent. THanks.
Installation Method: Scratch Install on centos 6.9 final, method followed (Document provided by Ray Solomon. Asterisk version 11.22.0-vici). 1x Dedicated Server from Oneprovider.
Server Specs;
CPU 2x Intel Xeon E5-2620 - 2 GHz - 6 core(s)
RAM
128GB - DDR4
STORAGE
3x 120GB (SSD SATA)
BANDWIDTH
Unmetered @ 1Gbps
-------------------------------
Hello experts, I have recently migrated database from a running server to a new server. (Both server specs and installation method is same). I followed all the steps required to get things going on the new server as i have done this a lot of times previously without facing any major issues and everything seems to start smoothly after migration. But this time things are acting a bit wierd. Initially everything seems to be normal like i can login, i'm also able to register the carrier, get the softphone online and also able to dial manual calls via Eyebeam registered to a vicidial phone ext to thension. The weirdness starts when we login to the campaign I see calls connecting and going into the survey but then nothing happens, like I am not able to see a single call entering the survey or the call menu(IVR). The call flow is like number gets answered enters the 8373 survey where it plays sip-silence for 1 second and enters a call menu where it gets the treatment, hears the recording and follows the instructions. Now the point is when the call enters the 8373 survey i see it on the asterisk console but unable to see any calls in survey or IVR on Realtime screen. I tried Predictive Dialer extension 8369 and 8368 still i dont see any LIVE calls towards the available agent. below is a snippet from asterisk.
[Oct 11 02:27:06] -- Executing [8368@default:1] Playback("SIP/usa-0000021a", "sip-silence") in new stack
[Oct 11 02:27:06] -- Executing [h@default:1] AGI("Local/52317619256919696@default-0000025f;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----0") in new stack
[Oct 11 02:27:06] -- <SIP/usa-0000021a> Playing 'sip-silence.gsm' (language 'en')
[Oct 11 02:27:06] > 0x7f83ac016d40 -- Probation passed - setting RTP source address to 216.221.155.204:59578
[Oct 11 02:27:07] -- Executing [8368@default:2] AGI("SIP/usa-0000021a", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 11 02:27:07] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=EBP3))
[Oct 11 02:27:07] -- <SIP/usa-0000021a>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 11 02:27:07] -- Executing [8368@default:3] AGI("SIP/usa-0000021a", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Oct 11 02:27:07] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 11 02:27:07] -- <SIP/usa-0000021a>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 11 02:27:07] -- Executing [8368@default:4] AGI("SIP/usa-0000021a", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Oct 11 02:27:07] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 11 02:27:07] -- <SIP/usa-0000021a>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 11 02:27:07] -- Executing [8368@default:5] Hangup("SIP/usa-0000021a", "") in new stack
[Oct 11 02:27:07] == Spawn extension (default, 8368, 5) exited non-zero on 'SIP/usa-0000021a'
[Oct 11 02:27:07] -- Executing [h@default:1] AGI("SIP/usa-0000021a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Oct 11 02:27:07] -- <SIP/usa-0000021a>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 11 02:27:08] -- <Local/52317619256919696@default-0000025f;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --20-----0 completed, returning 0
can u please help me rectify this issue or identify what is making the calls hangup? above snippet is when i switched it to ext 8368 to make any available call land on the agent but nothing happened instead the voip minutes are being used like normal but i dont know where the calls are going or whats wrong. please get me a fix or advice for this as its really urgent. THanks.