ViciDial On Hook rings but then cuts off

[penfold15]

Hi,

I am running goautodial 2.1 including bundled software, no extras.

I have setup a phone that will allow on hook dialling. I have set this up on a softphone and Hard phones.

The problem is that when the agent receives an incoming call the phone rings but when the agent answers it cuts off and then rings again in a few minutes and does the same thing again.

I have added the ring_all functions into the extensions.php

Any suggestions???

[williamconley]

1) Welcome to the Party!

2) Newbie suggestions:

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

Worthy of note: the only system information you listed was your installer version. Nothing about Vicidial or asterisk at all ... (ie: on the line above, you successfully filled in the First Item and the 6th, plus we can assume the 4th and 5th ...). Perhaps you could at least fill in # 2 which is required

3) Before you attempt "on-hook agents" you should try starting at the first page of the manual and working your way to the end. After you have completed the tutorials to the point that you have a functional dialer (using regular agents, not on-hook ones ...), you would then make that single change (from live agents to on-hook) and hope not to break anything with that small change.

4) If you bump into a problem, give us the Page # and paragraph (and version!) of the manager's manual you are using and a sample of the "flaw" (such as asterisk command line out put with the error ... or the error from an agent screen ... if you have it). Then we can help you get this online in an orderly fashion.

5) Happy Hunting!

[rgwilliams20]

1) Welcome to the Party!

2) Newbie suggestions:

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on , both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

Worthy of note: the only system information you listed was your installer version. Nothing about Vicidial or asterisk at all ... (ie: on the line above, you successfully filled in the First Item and the 6th, plus we can assume the 4th and 5th ...). Perhaps you could at least fill in # 2 which is required

3) Before you attempt "on-hook agents" you should try starting at the first page of the manual and working your way to the end. After you have completed the tutorials to the point that you have a functional dialer (using regular agents, not on-hook ones ...), you would then make that single change (from live agents to on-hook) and hope not to break anything with that small change.

4) If you bump into a problem, give us the Page # and paragraph (and version!) of the manager's manual you are using and a sample of the "flaw" (such as asterisk command line out put with the error ... or the error from an agent screen ... if you have it). Then we can help you get this online in an orderly fashion.

5) Happy Hunting!

Thank you for your reply, this is the information below:

Goautodial 2.1 from goautodial-ce-2.1-final.iso | Vicidial 2.4-309a Build 110430-1642 | Asterisk Not sure how to find but its the default from the iso| Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel(R) Core(TM) i7-2600 CPU @ 3.40GHz

We have run this dialler for around 18 months with up to 40 agents using it at any one time. These were connecting using a variation of SIP and IAX accounts. These have all worked without a problem. I have been working on this for a few hours and have found that the problem is "very" intermitant", the agent will receive the call, answer and then it will connect successfully, however for no apparent reason the next time you try the phone will ring but then when answered immediately disconnect.

I have tried this using softphones (zoiper and x-lite) and also sip phones (aastra 6751i) however I get the same problem on all of these, I can only assume that this has something to do with the conference feature within asterisk.

I have also added the following into extensions.conf

; these are used for the ring_all function in VICIDIAL
exten => _8331*.,1,Playback(sip-silence)
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,Hangup

This is what I am getting in the asterisk logs


[Mar 7 08:58:19] DEBUG[19549] chan_iax2.c: prepending 2 to prefs
[Mar 7 08:58:19] VERBOSE[19549] logger.c: [Mar 7 08:58:19] -- Called BM010
[Mar 7 08:58:19] VERBOSE[3669] logger.c: [Mar 7 08:58:19] -- Call accepted by 192.168.23.109 (format gsm)
[Mar 7 08:58:19] VERBOSE[3669] logger.c: [Mar 7 08:58:19] -- Format for call is gsm
[Mar 7 08:58:19] VERBOSE[19549] logger.c: [Mar 7 08:58:19] -- IAX2/BM010-8596 is ringing
[Mar 7 08:58:20] VERBOSE[19549] logger.c: [Mar 7 08:58:20] -- IAX2/BM010-8596 answered Local/192*168*023*026*010@default-6ce1,2
[Mar 7 08:58:20] VERBOSE[19548] logger.c: [Mar 7 08:58:20] > Channel Local/192*168*023*026*010@default-6ce1,1 was answered.
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Executing [8331*12*Y3070858180001148693*AdminSOT*010@default:1] Playback("Local/192*168*023*026*010@default-6ce1,1", "sip-silence") in new stack
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (language 'en')
[Mar 7 08:58:20] WARNING[19563] file.c: Unexpected control subclass '-1'
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Executing [8331*12*Y3070858180001148693*AdminSOT*010@default:2] AGI("Local/192*168*023*026*010@default-6ce1,1", "agi-VDAD_RINGALL.agi|8331*12*Y3070858180001148693*AdminSOT*010") in new stack
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
[Mar 7 08:58:20] VERBOSE[19549] logger.c: [Mar 7 08:58:20] -- Executing [h@default:1] DeadAGI("Local/192*168*023*026*010@default-6ce1,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----1-----0") in new stack
[Mar 7 08:58:20] VERBOSE[19549] logger.c: [Mar 7 08:58:20] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---1-----0 completed, returning 0
[Mar 7 08:58:20] VERBOSE[19549] logger.c: [Mar 7 08:58:20] == Spawn extension (default, 010, 1) exited non-zero on 'Local/192*168*023*026*010@default-6ce1,2'
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- AGI Script agi-VDAD_RINGALL.agi completed, returning 0
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Executing [8331*12*Y3070858180001148693*AdminSOT*010@default:3] AGI("IAX2/BM010-8596", "agi-VDAD_RINGALL.agi|8331*12*Y3070858180001148693*AdminSOT*010") in new stack
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] ERROR[19563] utils.c: write() returned error: Broken pipe
[Mar 7 08:58:20] ERROR[19563] utils.c: write() returned error: Broken pipe
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- AGI Script agi-VDAD_RINGALL.agi completed, returning 0
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Executing [192*168*023*026*78600051@default:1] Goto("IAX2/BM010-8596", "default|78600051|1") in new stack
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Goto (default,78600051,1)
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Executing [78600051@default:1] MeetMe("IAX2/BM010-8596", "8600051|Fq") in new stack
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] == Parsing '/etc/asterisk/meetme.conf': [Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] Found
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] Found
[Mar 7 08:58:20] VERBOSE[19563] logger.c: [Mar 7 08:58:20] -- Created MeetMe conference 1023 for conference '8600051'
[Mar 7 08:58:20] VERBOSE[19537] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] VERBOSE[19537] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] VERBOSE[19537] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] VERBOSE[19537] logger.c: [Mar 7 08:58:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:20] ERROR[19537] utils.c: write() returned error: Broken pipe
[Mar 7 08:58:20] VERBOSE[19537] logger.c: [Mar 7 08:58:20] -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Mar 7 08:58:20] VERBOSE[19537] logger.c: [Mar 7 08:58:20] -- Executing [192*168*023*026*8600051@default:1] Goto("SIP/voiceflexsip2-0000000c", "default|8600051|1") in new stack
[Mar 7 08:58:20] VERBOSE[19537] logger.c: [Mar 7 08:58:20] -- Goto (default,8600051,1)
[Mar 7 08:58:20] VERBOSE[19537] logger.c: [Mar 7 08:58:20] -- Executing [8600051@default:1] MeetMe("SIP/voiceflexsip2-0000000c", "8600051|F") in new stack
[Mar 7 08:58:22] VERBOSE[19537] logger.c: [Mar 7 08:58:22] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/voiceflexsip2-0000000c'
[Mar 7 08:58:22] VERBOSE[19537] logger.c: [Mar 7 08:58:22] -- Executing [h@default:1] DeadAGI("SIP/voiceflexsip2-0000000c", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Mar 7 08:58:22] VERBOSE[19537] logger.c: [Mar 7 08:58:22] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Mar 7 08:58:22] VERBOSE[19548] logger.c: [Mar 7 08:58:22] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 7 08:58:27] VERBOSE[19563] logger.c: [Mar 7 08:58:27] == Spawn extension (default, 78600051, 1) exited non-zero on 'IAX2/BM010-8596'
[Mar 7 08:58:27] VERBOSE[19563] logger.c: [Mar 7 08:58:27] -- Executing [h@default:1] DeadAGI("IAX2/BM010-8596", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Mar 7 08:58:27] VERBOSE[19563] logger.c: [Mar 7 08:58:27] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Mar 7 08:58:27] VERBOSE[19563] logger.c: [Mar 7 08:58:27] -- Hungup 'IAX2/BM010-8596'
[Mar 7 08:58:34] VERBOSE[19665] logger.c: [Mar 7 08:58:34] -- Executing [01782433122@trunkinbound:1] AGI("SIP/voiceflexsip2-0000000d", "agi-DID_route.agi") in new stack
[Mar 7 08:58:34] VERBOSE[19665] logger.c: [Mar 7 08:58:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Mar 7 08:58:34] ERROR[19665] utils.c: write() returned error: Broken pipe
[Mar 7 08:58:34] ERROR[19665] utils.c: write() returned error: Broken pipe
[Mar 7 08:58:34] VERBOSE[19665] logger.c: [Mar 7 08:58:34] -- AGI Script agi-DID_route.agi completed, returning 0
[Mar 7 08:58:34] VERBOSE[19665] logger.c: [Mar 7 08:58:34] -- Executing [99909*18***DID@default:1] Answer("SIP/voiceflexsip2-0000000d", "") in new stack
[Mar 7 08:58:34] VERBOSE[19665] logger.c: [Mar 7 08:58:34] -- Executing [99909*18***DID@default:2] AGI("SIP/voiceflexsip2-0000000d", "agi-VDAD_ALL_inbound.agi") in new stack
[Mar 7 08:58:34] VERBOSE[19665] logger.c: [Mar 7 08:58:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Mar 7 08:58:34] VERBOSE[19665] logger.c: [Mar 7 08:58:34] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:34] VERBOSE[19665] logger.c: [Mar 7 08:58:34] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:36] VERBOSE[19665] logger.c: [Mar 7 08:58:36] -- Started music on hold, class '054', on SIP/voiceflexsip2-0000000d
[Mar 7 08:58:39] VERBOSE[19708] logger.c: [Mar 7 08:58:39] == Parsing '/etc/asterisk/manager.conf': [Mar 7 08:58:39] VERBOSE[19708] logger.c: [Mar 7 08:58:39] Found
[Mar 7 08:58:39] VERBOSE[19708] logger.c: [Mar 7 08:58:39] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 7 08:58:39] VERBOSE[19709] logger.c: [Mar 7 08:58:39] -- Executing [192*168*023*026*010@default:1] Goto("Local/192*168*023*026*010@default-a2bc,2", "default|010|1") in new stack
[Mar 7 08:58:39] VERBOSE[19709] logger.c: [Mar 7 08:58:39] -- Goto (default,010,1)
[Mar 7 08:58:39] VERBOSE[19709] logger.c: [Mar 7 08:58:39] -- Executing [010@default:1] Dial("Local/192*168*023*026*010@default-a2bc,2", "IAX2/BM010|60|") in new stack
[Mar 7 08:58:39] DEBUG[19709] chan_iax2.c: prepending 2 to prefs
[Mar 7 08:58:39] VERBOSE[19709] logger.c: [Mar 7 08:58:39] -- Called BM010
[Mar 7 08:58:40] VERBOSE[3628] logger.c: [Mar 7 08:58:40] -- Call accepted by 192.168.23.109 (format gsm)
[Mar 7 08:58:40] VERBOSE[3628] logger.c: [Mar 7 08:58:40] -- Format for call is gsm
[Mar 7 08:58:40] VERBOSE[19709] logger.c: [Mar 7 08:58:40] -- IAX2/BM010-1949 is ringing
[Mar 7 08:58:41] VERBOSE[19709] logger.c: [Mar 7 08:58:41] -- IAX2/BM010-1949 answered Local/192*168*023*026*010@default-a2bc,2
[Mar 7 08:58:41] VERBOSE[19708] logger.c: [Mar 7 08:58:41] > Channel Local/192*168*023*026*010@default-a2bc,1 was answered.
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Executing [8331*13*Y3070858340001148694*AdminSOT*010@default:1] Playback("Local/192*168*023*026*010@default-a2bc,1", "sip-silence") in new stack
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Playing 'sip-silence' (language 'en')
[Mar 7 08:58:41] WARNING[19721] file.c: Unexpected control subclass '-1'
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Executing [8331*13*Y3070858340001148694*AdminSOT*010@default:2] AGI("Local/192*168*023*026*010@default-a2bc,1", "agi-VDAD_RINGALL.agi|8331*13*Y3070858340001148694*AdminSOT*010") in new stack
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- AGI Script agi-VDAD_RINGALL.agi completed, returning 0
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Executing [8331*13*Y3070858340001148694*AdminSOT*010@default:3] AGI("Local/192*168*023*026*010@default-a2bc,1", "agi-VDAD_RINGALL.agi|8331*13*Y3070858340001148694*AdminSOT*010") in new stack
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- AGI Script agi-VDAD_RINGALL.agi completed, returning 0
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Executing [8331*13*Y3070858340001148694*AdminSOT*010@default:4] Hangup("Local/192*168*023*026*010@default-a2bc,1", "") in new stack
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] == Spawn extension (default, 8331*13*Y3070858340001148694*AdminSOT*010, 4) exited non-zero on 'Local/192*168*023*026*010@default-a2bc,1'
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- Executing [h@default:1] DeadAGI("Local/192*168*023*026*010@default-a2bc,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Mar 7 08:58:41] VERBOSE[19721] logger.c: [Mar 7 08:58:41] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Mar 7 08:58:41] VERBOSE[19709] logger.c: [Mar 7 08:58:41] -- Executing [h@default:1] DeadAGI("Local/192*168*023*026*010@default-a2bc,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----2-----0") in new stack
[Mar 7 08:58:41] VERBOSE[19709] logger.c: [Mar 7 08:58:41] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---2-----0 completed, returning 0
[Mar 7 08:58:41] VERBOSE[19709] logger.c: [Mar 7 08:58:41] -- Hungup 'IAX2/BM010-1949'
[Mar 7 08:58:41] VERBOSE[19709] logger.c: [Mar 7 08:58:41] == Spawn extension (default, 010, 1) exited non-zero on 'Local/192*168*023*026*010@default-a2bc,2'
[Mar 7 08:58:41] VERBOSE[19665] logger.c: [Mar 7 08:58:41] -- Stopped music on hold on SIP/voiceflexsip2-0000000d
[Mar 7 08:58:41] VERBOSE[19665] logger.c: [Mar 7 08:58:41] -- Started music on hold, class '054', on SIP/voiceflexsip2-0000000d
[Mar 7 08:58:43] VERBOSE[19708] logger.c: [Mar 7 08:58:43] == Manager 'sendcron' logged off from 127.0.0.1






Any assistance would be great

[williamconley]

Asterisk Not sure how to find but its the default from the iso

Code: Select all

asterisk -R

Then "exit"

Change from ringall to another strategy. You are likely creating a race condition. All agent phones ring .. first to answer wins and the rest have to hang up the phone and wait for another call. That's how ringall works.

Also consider increasing the ring time.