I had previously created this post. Callers lose there place in queue
PBXact 15.0.17.55
Version 12.7.8-2107-3.sng7
Ended up setting up a PBxact UC60 from scratch and they still have the same problem!
They have crazy high call volume and long waits in the queues, so their customers are really upset when it says there are number 1 in the queue, then either the go back to a higher number, like 5 in queue or the calls drops all together.
Still getting complaints about being down to caller #1 then the music will quit, the call drops, then they can start over. Had a complaint yesterday and already this morning about that.
How do I do further tracing to determine the cause of this?
grep did exactly what you told it to do. You told it to look for call 0000070a in the file dated the october 19th, when the prior search clearly told you it was in a file dated october 20th.
That is not a call from yesterday, looking at old log from a completely different system will not troubleshoot anything on your new system.
Sorry, I’m not well versed in grep, or other linux commands, other than what I have learned in freebpx. I was attempting to look for the file from the 19th, since that’s when they call queue drop had happened. I’m working on a current call queue drop now.
Here’s the result of the first search https://pastebin.freepbx.org/view/bca16bed
When I did just entered the /asterisk/full it gets results,so I think I have what I need, but when I do the /full-20211123
it doesn’t show any results.
When I searched for the system id of the call that dropped out of queue, I get the results below, which I had in the first pastebin link. From the providing great sip debug link, I understood the Asterisk Call-ID, which is C-00005c85 , is what I needed to filter the call trace. Am I not understanding that correctly?
[root@phonesystem ~]# grep 1637688260.1658782 /var/log/asterisk/full*
/var/log/asterisk/full:[2021-11-23 11:24:20] VERBOSE[4890][C-00005c85] pbx.c: Executing [3046611559@from-trunk:22] Set(“SIP/Voip_Innovations_In-0002fb1f”, “__CRM_LINKEDID=1637688260.1658782”) in new stack
/var/log/asterisk/full:[2021-11-23 11:24:21] VERBOSE[4890][C-00005c85] res_agi.c: sangomacrm.agi,true: LINKEDID: 1637688260.1658782
/var/log/asterisk/full:[2021-11-23 11:24:37] VERBOSE[4890][C-00005c85] pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/Voip_Innovations_In-0002fb1f”, “TOUCH_MONITOR=1637688260.1658782”) in new stack
/var/log/asterisk/full:[2021-11-23 11:24:48] VERBOSE[4890][C-00005c85] pbx.c: Executing [802@ext-queues:41] QueueLog(“SIP/Voip_Innovations_In-0002fb1f”, “802,1637688260.1658782,NONE,DID,3046611559”) in new stack
[root@phonesystem ~]# grep C-00005c85 /var/log/asterisk/full-20211124
grep: /var/log/asterisk/full-20211124: No such file or directory
[root@phonesystem ~]#
Think I’m seeing the wonders now. Must be something in how the log files are created by date,the one for the 24th includes data from the 23rd. So I ran that grep search in the full-20211124 log this morning, but it still gives me the same info as my first pastebin link, but here it is again. Automatic Pastebin from Sangoma OS 7 - FreePBX Pastebin
The info lines up with the time stamp from the CDR reports. At 11:56 is when the call drops. I see these errors
Line 1527441: [2021-11-23 11:56:44] VERBOSE[5500][C-00005c85] pbx.c: Executing [s@macro-exten-vm:5] Set(“Local/102@from-queue-000b212e;2”, “RT=25”) in new stack
Line 1527442: [2021-11-23 11:56:44] WARNING[5500][C-00005c85] chan_sip.c: This function can only be used on SIP channels.
Line 1527443: [2021-11-23 11:56:44] VERBOSE[5500][C-00005c85] pbx.c: Executing [s@macro-exten-vm:6] ExecIf(“Local/102@from-queue-000b212e;2”, “0?Macro(vm,102,DIRECTDIAL,)”) in new stack
Line 1527444: [2021-11-23 11:56:44] WARNING[5500][C-00005c85] chan_sip.c: This function can only be used on SIP channels.
Line 1527445: [2021-11-23 11:56:44] WARNING[5500][C-00005c85] chan_sip.c: This function can only be used on SIP channels.
Line 1527446: [2021-11-23 11:56:44] VERBOSE[5500][C-00005c85] pbx.c: Executing [s@macro-exten-vm:7] ExecIf(“Local/102@from-queue-000b212e;2”, “0?MacroExit()”) in new stack
Line 1527447: [2021-11-23 11:56:44] WARNING[5500][C-00005c85] chan_sip.c: This function can only be used on SIP channels.
Line 1527448: [2021-11-23 11:56:44] ERROR[5500][C-00005c85] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
Line 1527449: [2021-11-23 11:56:44] VERBOSE[5500][C-00005c85] pbx.c: Executing [s@macro-exten-vm:8] ExecIf(“Local/102@from-queue-000b212e;2”, “0?Macro(vm,102,DIRECTDIAL,)”) in new stack
Line 1527450: [2021-11-23 11:56:44] ERROR[5500][C-00005c85] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
Line 1527451: [2021-11-23 11:56:44] ERROR[5500][C-00005c85] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
Line 1527452: [2021-11-23 11:56:44] VERBOSE[5500][C-00005c85] pbx.c: Executing [s@macro-exten-vm:9] ExecIf(“Local/102@from-queue-000b212e;2”, “0?MacroExit()”) in new stack
Line 1527453: [2021-11-23 11:56:44] ERROR[5500][C-00005c85] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
Line 1527454: [2021-11-23 11:56:44] VERBOSE[5500][C-00005c85] pbx.c: Executing [s@macro-exten-vm:10] Gosub(“Local/102@from-queue-000b212e;2”, “sub-record-check,s,1(exten,102,dontcare)”) in new stack
So I changed the sip channel driver from both to chansip since that’s what is being used. But I’m not sure if that’s related or not.
The complaint is this " The caller got cut off 4 times when he got You are now the next person in line then the music would play for a minute or so then stop. He would hear a couple of clicks and the call was terminated."
Your agents are not answering when their device is ringing. Fix the user issue.
As for everything else, Check your queue timeouts, and failover settings. Make sure nothing is going “back” into the queue, as that would do exactly what you are describing.
I like sorvani’s answer. It is real world, to the point. If you are running a call center your agents should be answering the calls. 32 minutes is not a call center.
