Moderators: gerski, enjay, williamconley, Op3r, Staydog, gardo, mflorell, MJCoate, mcargile, Kumba, Michael_N
limits of my server are being tested
mubeen wrote:Thank you for detailed reply. Just for testing purpose I went ahead and installed VICIBOX 7.0.4 (downloaded from vicidial archive) on all machines with cluster configuration.
What exactly I did was
Run vicibox and installed OS using os-install, then using vicidial-install I installed DB then Archive, then web and telephony in last. After installation I created campaign and created users. I also added all 3 servers in GUI from add server option in webserver.
Now when I login as an agent, connecting calls come on eyebeam, It does say you are currently only....but after like 10-15 sec session disconnects
What limits? I presume there's an error somewhere or long wait times or something is RED? Without that information it's hard to help without writing a book to cover every scenario (which is not likely today, but Kumba Stepped Up which is cool).
Use the vicidial manager's manual available on EFLO.net and start at the first page. Post here the first error you bump into, the page/line you're on, and what happened. Also post what you Expected to happen and any relevent configuration information. Don't skip any pages. Start at the front and keep going until everything you need works.
In this case, it would also be very useful to post the asterisk CLI from a single agent login. From inception to termination, one full call, but don't have anything else going on at the same time (there should only be a couple dozen lines of code, not 3000 lines of unrelated output, lol).
Please also remember to Always post your Full Vicidial Version with Build. Essential for troubleshooting both now and months from now when someone else reads this post.
[Oct 26 17:11:15] WARNING[1441]: chan_sip.c:4101 retrans_pkt: Timeout on 2135780793-357499114-1980619025 on non-critical invite transaction.
[Oct 26 17:11:16] WARNING[1441]: chan_sip.c:4101 retrans_pkt: Timeout on 3734824d544c19b098141eb2033883d9 on non-critical invite transaction.
[Oct 26 17:11:35] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 17:11:35] == Using SIP RTP CoS mark 5
[Oct 26 17:11:37] > Channel SIP/1001-0000001a was answered
[Oct 26 17:11:37] -- Executing [8600052@default:1] MeetMe("SIP/1001-0000001a", "8600052,F") in new stack
[Oct 26 17:11:37] == Parsing '/etc/asterisk/meetme.conf': Found
[Oct 26 17:11:37] == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[Oct 26 17:11:37] -- Created MeetMe conference 1023 for conference '8600052'
[Oct 26 17:11:37] -- <SIP/1001-0000001a> Playing 'conf-onlyperson.gsm' (language 'en')
[Oct 26 17:11:37] > 0x7fe070019f80 -- Probation passed - setting RTP source address to 182.176.118.54:6400
[Oct 26 17:11:38] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 17:11:46] WARNING[1441]: chan_sip.c:4101 retrans_pkt: Timeout on 1a22f1be9318a6608d7624a3b5b9dfad on non-critical invite transaction.
[Oct 26 17:12:01] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 17:12:01] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 17:12:01] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 17:12:06] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 17:12:06] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 17:12:11] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 17:12:15] WARNING[1441]: chan_sip.c:4101 retrans_pkt: Timeout on 597b745fbff2f61523f4d1c495037c56 on non-critical invite transaction.
it has already reached the limit of agents it can handle
- Code: Select all
Playing 'conf-onlyperson
Ok ... how do you know? Is there a popup window saying "You've exceeded the number of agents this server can handle!" or am I missing some information? (I'll bet it's the latter)
- Code: Select all
Playing 'conf-onlyperson
I face call issues.
mubeen wrote:server web interface takes time to reach
nano /etc/apache2/server-tuning.conf
# prefork MPM
<IfModule prefork.c>
# number of server processes to start
# http://httpd.apache.org/docs/2.2/mod/mpm_common.html#startservers
StartServers 450
# minimum number of server processes which are kept spare
# http://httpd.apache.org/docs/2.2/mod/prefork.html#minspareservers
MinSpareServers 250
# maximum number of server processes which are kept spare
# http://httpd.apache.org/docs/2.2/mod/prefork.html#maxspareservers
MaxSpareServers 500
# highest possible MaxClients setting for the lifetime of the Apache process.
# http://httpd.apache.org/docs/2.2/mod/mpm_common.html#serverlimit
ServerLimit 768
# maximum number of server processes allowed to start
# http://httpd.apache.org/docs/2.2/mod/mpm_common.html#maxclients
MaxClients 768
# maximum number of requests a server process serves
# http://httpd.apache.org/docs/2.2/mod/mpm_common.html#maxrequestsperchild
MaxRequestsPerChild 1000
</IfModule>
nano /etc/apache2/default-server.conf
# Load the status module for http://XXX.poundteam.com/server-status
LoadModule status_module /usr/lib64/apache2/mod_status.so
ExtendedStatus On
<Location /server-status>
SetHandler server-status
</Location>
service apache2 restart
http://xx.xx.xx.xx/server-status
mubeen wrote:Agents are active but being shown paused on admin screen
mubeen wrote:using press-1 campaign giving issues like for VOIP number, call is being transferred to agents but drops for real number with buzz sound (which I never was able to resolve)
mubeen wrote:This is the main reason I'm moving for cluster setup. I do have to run the setup (to start with for new campaign) for 10 agents with dial level 100 or so.
mubeen wrote:using press-1 campaign giving issues like for VOIP number, call is being transferred to agents but drops for real number with buzz sound (which I never was able to resolve)
I didn't get that at all. But it sounds like you may need to change carriers. Please describe in more detail (and try with another carrier to see if your problem magically disappears).
[Oct 1 09:42:05] VERBOSE[27262][C-0009b15f] pbx.c: [Oct 1 09:42:05] -- Executing [91305xxxxxxx@default:1] AGI("Local/91305xxxxxxx@default-0008257c;2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 1 09:42:05] VERBOSE[27262][C-0009b15f] res_agi.c: [Oct 1 09:42:05] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=HS_IVR_2))
[Oct 1 09:42:05] VERBOSE[27262][C-0009b15f] res_agi.c: [Oct 1 09:42:05] -- <Local/91305xxxxxxx@default-0008257c;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 1 09:42:05] VERBOSE[27262][C-0009b15f] pbx.c: [Oct 1 09:42:05] -- Executing [91305xxxxxxx@default:2] Dial("Local/91305xxxxxxx@default-0008257c;2", "sip/1305xxxxxxx@Carrier,55,tTor") in new stack
[Oct 1 09:42:05] VERBOSE[27262][C-0009b15f] netsock2.c: [Oct 1 09:42:05] == Using SIP RTP CoS mark 5
[Oct 1 09:42:05] VERBOSE[27262][C-0009b15f] app_dial.c: [Oct 1 09:42:05] -- Called sip/1305xxxxxxx@Carrier
[Oct 1 09:42:06] VERBOSE[27262][C-0009b15f] app_dial.c: [Oct 1 09:42:06] -- SIP/Carrier-000699c0 is ringing
[Oct 1 09:42:06] VERBOSE[20030][C-0009b15f] res_rtp_asterisk.c: [Oct 1 09:42:06] > 0x7f805000ab30 -- Strict RTP learning after remote address set to: 169.132.xxx.xxx:23666
[Oct 1 09:42:06] VERBOSE[27262][C-0009b15f] app_dial.c: [Oct 1 09:42:06] -- SIP/Carrier-000699c0 is making progress passing it to Local/91305xxxxxxx@default-0008257c;2
[Oct 1 09:42:06] VERBOSE[27262][C-0009b15f] res_rtp_asterisk.c: [Oct 1 09:42:06] > 0x7f805000ab30 -- Strict RTP switching to RTP remote address 169.132.xxx.xxx:23666 as source
[Oct 1 09:42:08] VERBOSE[27262][C-0009b15f] res_rtp_asterisk.c: [Oct 1 09:42:08] > 0x7f805000ab30 -- Strict RTP learning complete - Locking on source address 169.132.xxx.xxx:23666
[Oct 1 09:42:16] VERBOSE[27262][C-0009b15f] app_dial.c: [Oct 1 09:42:16] -- SIP/Carrier-000699c0 answered Local/91305xxxxxxx@default-0008257c;2
[Oct 1 09:42:16] VERBOSE[27261][C-0009b15f] pbx.c: [Oct 1 09:42:16] > Channel Local/91305xxxxxxx@default-0008257c;1 was answered
[Oct 1 09:42:16] VERBOSE[27504][C-0009b15f] pbx.c: [Oct 1 09:42:16] -- Executing [8366@default:1] Playback("SIP/Carrier-000699c0", "sip-silence") in new stack
[Oct 1 09:42:16] VERBOSE[27262][C-0009b15f] pbx.c: [Oct 1 09:42:16] -- Executing [h@default:1] AGI("Local/91305xxxxxxx@default-0008257c;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----11-----0") in new stack
[Oct 1 09:42:16] VERBOSE[27504][C-0009b15f] file.c: [Oct 1 09:42:16] -- <SIP/Carrier-000699c0> Playing 'sip-silence.gsm' (language 'en')
[Oct 1 09:42:16] VERBOSE[27504][C-0009b15f] pbx.c: [Oct 1 09:42:16] -- Executing [8366@default:2] AGI("SIP/Carrier-000699c0", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 1 09:42:16] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:16] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=HS_IVR_2))
[Oct 1 09:42:16] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:16] -- <SIP/Carrier-000699c0>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 1 09:42:16] VERBOSE[27504][C-0009b15f] pbx.c: [Oct 1 09:42:16] -- Executing [8366@default:3] AGI("SIP/Carrier-000699c0", "agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB") in new stack
[Oct 1 09:42:16] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:16] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 1 09:42:16] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:16] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20181001-094216_305xxxxxxx)
[Oct 1 09:42:16] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:16] -- <SIP/Carrier-000699c0> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 1 09:42:16] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:16] -- <SIP/Carrier-000699c0> Playing 'OUT-8khz.slin' (escape_digits=123) (sample_offset 0) (language 'en')
[Oct 1 09:42:17] VERBOSE[27262][C-0009b15f] res_agi.c: [Oct 1 09:42:17] -- <Local/91305xxxxxxx@default-0008257c;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----11-----0 completed, returning 0
[Oct 1 09:42:17] VERBOSE[27262][C-0009b15f] pbx.c: [Oct 1 09:42:17] == Spawn extension (default, 91305xxxxxxx, 2) exited non-zero on 'Local/91305xxxxxxx@default-0008257c;2'
[Oct 1 09:42:27] DTMF[27504][C-0009b15f] channel.c: DTMF begin '1' received on SIP/Carrier-000699c0
[Oct 1 09:42:27] DTMF[27504][C-0009b15f] channel.c: DTMF begin ignored '1' on SIP/Carrier-000699c0
[Oct 1 09:42:27] DTMF[27504][C-0009b15f] channel.c: DTMF end '1' received on SIP/Carrier-000699c0, duration 180 ms
[Oct 1 09:42:27] DTMF[27504][C-0009b15f] channel.c: DTMF end passthrough '1' on SIP/Carrier-000699c0
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:33] -- <SIP/Carrier-000699c0> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:33] -- <SIP/Carrier-000699c0> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:33] -- <SIP/Carrier-000699c0> Playing 'buzz.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:33] -- <SIP/Carrier-000699c0>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] pbx.c: [Oct 1 09:42:33] == Spawn extension (default, 8366, 3) exited non-zero on 'SIP/Carrier-000699c0'
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] pbx.c: [Oct 1 09:42:33] -- Executing [h@default:1] AGI("SIP/Carrier-000699c0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:33] -- <SIP/Carrier-000699c0>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Oct 1 07:54:32] VERBOSE[7553][C-0009ad94] pbx.c: [Oct 1 07:54:32] -- Executing [91909xxxxxxx@default:1] AGI("Local/91909xxxxxxx@default-000821c6;2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 1 07:54:32] VERBOSE[7553][C-0009ad94] res_agi.c: [Oct 1 07:54:32] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=HS_IVR_2))
[Oct 1 07:54:32] VERBOSE[7553][C-0009ad94] res_agi.c: [Oct 1 07:54:32] -- <Local/91909xxxxxxx@default-000821c6;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 1 07:54:32] VERBOSE[7553][C-0009ad94] pbx.c: [Oct 1 07:54:32] -- Executing [91909xxxxxxx@default:2] Dial("Local/91909xxxxxxx@default-000821c6;2", "sip/1909xxxxxxx@Carrier,55,tTor") in new stack
[Oct 1 07:54:32] VERBOSE[7553][C-0009ad94] netsock2.c: [Oct 1 07:54:32] == Using SIP RTP CoS mark 5
[Oct 1 07:54:32] VERBOSE[7553][C-0009ad94] app_dial.c: [Oct 1 07:54:32] -- Called sip/1909xxxxxxx@Carrier
[Oct 1 07:54:36] VERBOSE[20030][C-0009ad94] res_rtp_asterisk.c: [Oct 1 07:54:36] > 0x7f8030014880 -- Strict RTP learning after remote address set to: 169.132.xxx.xxx:20896
[Oct 1 07:54:36] VERBOSE[7553][C-0009ad94] app_dial.c: [Oct 1 07:54:36] -- SIP/Carrier-00069619 is making progress passing it to Local/91909xxxxxxx@default-000821c6;2
[Oct 1 07:54:36] VERBOSE[7553][C-0009ad94] res_rtp_asterisk.c: [Oct 1 07:54:36] > 0x7f8030014880 -- Strict RTP switching to RTP remote address 169.132.xxx.xxx:20896 as source
[Oct 1 07:54:38] VERBOSE[7553][C-0009ad94] res_rtp_asterisk.c: [Oct 1 07:54:38] > 0x7f8030014880 -- Strict RTP learning complete - Locking on source address 169.132.xxx.xxx:20896
[Oct 1 07:54:38] VERBOSE[7553][C-0009ad94] app_dial.c: [Oct 1 07:54:38] -- SIP/Carrier-00069619 answered Local/91909xxxxxxx@default-000821c6;2
[Oct 1 07:54:38] VERBOSE[7552][C-0009ad94] pbx.c: [Oct 1 07:54:38] > Channel Local/91909xxxxxxx@default-000821c6;1 was answered
[Oct 1 07:54:38] VERBOSE[7563][C-0009ad94] pbx.c: [Oct 1 07:54:38] -- Executing [8366@default:1] Playback("SIP/Carrier-00069619", "sip-silence") in new stack
[Oct 1 07:54:38] VERBOSE[7553][C-0009ad94] pbx.c: [Oct 1 07:54:38] -- Executing [h@default:1] AGI("Local/91909xxxxxxx@default-000821c6;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----0") in new stack
[Oct 1 07:54:38] VERBOSE[7563][C-0009ad94] file.c: [Oct 1 07:54:38] -- <SIP/Carrier-00069619> Playing 'sip-silence.gsm' (language 'en')
[Oct 1 07:54:38] VERBOSE[7563][C-0009ad94] pbx.c: [Oct 1 07:54:38] -- Executing [8366@default:2] AGI("SIP/Carrier-00069619", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 1 07:54:38] VERBOSE[7563][C-0009ad94] res_agi.c: [Oct 1 07:54:38] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=HS_IVR_2))
[Oct 1 07:54:38] VERBOSE[7563][C-0009ad94] res_agi.c: [Oct 1 07:54:38] -- <SIP/Carrier-00069619>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 1 07:54:38] VERBOSE[7563][C-0009ad94] pbx.c: [Oct 1 07:54:38] -- Executing [8366@default:3] AGI("SIP/Carrier-00069619", "agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB") in new stack
[Oct 1 07:54:38] VERBOSE[7563][C-0009ad94] res_agi.c: [Oct 1 07:54:38] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 1 07:54:39] VERBOSE[7563][C-0009ad94] res_agi.c: [Oct 1 07:54:39] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20181001-075439_909xxxxxxx)
[Oct 1 07:54:39] VERBOSE[7563][C-0009ad94] res_agi.c: [Oct 1 07:54:39] -- <SIP/Carrier-00069619> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 1 07:54:39] VERBOSE[7563][C-0009ad94] res_agi.c: [Oct 1 07:54:39] -- <SIP/Carrier-00069619> Playing 'OUT-8khz.slin' (escape_digits=123) (sample_offset 0) (language 'en')
[Oct 1 07:54:39] VERBOSE[7553][C-0009ad94] res_agi.c: [Oct 1 07:54:39] -- <Local/91909xxxxxxx@default-000821c6;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----0 completed, returning 0
[Oct 1 07:54:39] VERBOSE[7553][C-0009ad94] pbx.c: [Oct 1 07:54:39] == Spawn extension (default, 91909xxxxxxx, 2) exited non-zero on 'Local/91909xxxxxxx@default-000821c6;2'
[Oct 1 07:54:43] DTMF[7563][C-0009ad94] channel.c: DTMF begin '1' received on SIP/Carrier-00069619
[Oct 1 07:54:43] DTMF[7563][C-0009ad94] channel.c: DTMF begin ignored '1' on SIP/Carrier-00069619
[Oct 1 07:54:44] DTMF[7563][C-0009ad94] channel.c: DTMF end '1' received on SIP/Carrier-00069619, duration 220 ms
[Oct 1 07:54:44] DTMF[7563][C-0009ad94] channel.c: DTMF end passthrough '1' on SIP/Carrier-00069619
[Oct 1 07:54:44] VERBOSE[7563][C-0009ad94] res_agi.c: [Oct 1 07:54:44] -- <SIP/Carrier-00069619>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 1 07:54:44] VERBOSE[7563][C-0009ad94] pbx.c: [Oct 1 07:54:44] -- Executing [144*076*003*247*8600051@default:1] Goto("SIP/Carrier-00069619", "default,8600051,1") in new stack
[Oct 1 07:54:44] VERBOSE[7563][C-0009ad94] pbx.c: [Oct 1 07:54:44] -- Goto (default,8600051,1)
[Oct 1 07:54:44] VERBOSE[7563][C-0009ad94] pbx.c: [Oct 1 07:54:44] -- Executing [8600051@default:1] MeetMe("SIP/Carrier-00069619", "8600051,F") in new stack
[Oct 1 07:54:51] VERBOSE[7563][C-0009ad94] pbx.c: [Oct 1 07:54:51] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/Carrier-00069619'
[Oct 1 07:54:51] VERBOSE[7563][C-0009ad94] pbx.c: [Oct 1 07:54:51] -- Executing [h@default:1] AGI("SIP/Carrier-00069619", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct 1 07:54:51] VERBOSE[7563][C-0009ad94] res_agi.c: [Oct 1 07:54:51] -- <SIP/Carrier-00069619>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
<SIP/Carrier-000699c0>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4
<SIP/Carrier-00069619>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
williamconley wrote:... If so, you should share the DTMF settings for them. Both ends if the one that failed was the VOIP number and you control that configuration as well.
[Oct 1 07:54:44] VERBOSE[7563][C-0009ad94] res_agi.c: [Oct 1 07:54:44] -- <SIP/Carrier-00069619>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 1 07:54:44] VERBOSE[7563][C-0009ad94] pbx.c: [Oct 1 07:54:44] -- Executing [144*076*003*247*8600051@default:1] Goto("SIP/Carrier-00069619", "default,8600051,1") in new stack
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:33] -- <SIP/Carrier-000699c0> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:33] -- <SIP/Carrier-000699c0> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:33] -- <SIP/Carrier-000699c0> Playing 'buzz.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] res_agi.c: [Oct 1 09:42:33] -- <SIP/Carrier-000699c0>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4
[Oct 1 09:42:33] VERBOSE[27504][C-0009b15f] pbx.c: [Oct 1 09:42:33] == Spawn extension (default, 8366, 3) exited non-zero on 'SIP/Carrier-000699c0'
And I'm using call menus for other campaigns, its working fine for inbound calls as well as outbound with PD
call menus for other campaigns and I did tested it for both VOIP and real numbers on same server
mubeen wrote:On further investigation, when I exported the report I found the status of all calls on which 1 was pressed was "Agent not available"
Agent Not Available
3 agents logged in and waiting for calls
mubeen wrote:How can I check this in logs? That agents are logged in in same Ingroup as these calls? I mean I can see the status in real time but not sure about logs
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