Survey Stopped Playing Audio

[Kinghtflyer]

ViciBox Redux 4.0.1
Preload ISO

I struggled and finally got a survey test campaign to work with all our prompts.
using several cell phones all was solid!
I then copied the campaign and loaded 20K leads
about a 100 calls in we determined no audio out for the callee.
The callee would only hear silence and then hangup.
Bummer.

I retested the original test campaign with cell phones, same issue.
was not able to determine what I broke so I reverted to newly loading from the ISO.
Now I cannot get survey outbound audio to work.

The owner of all files in var lib asterisk sounds is root.root, including the wav files I uploaded.
I feel that is the issue, despite everyone having Read perms.

All dtmf responses don't appear to be working either.

Can you help me figure out what stupid thing I am doing wrong?

I cannot post the CLI for your forum software wants me to remove all urls (Too spamy for a new user)

[williamconley]

1) Welcome to the party! (Sorry to hear the obvious frustration in your typing.)

2) 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

3) Before you make any changes to the survey or system, begin with a fresh installation and just use the sample Survey. Be sure it works completely. Change ONE thing (recording file?) and test again. Be sure everything works again. Keep doing this (one thing at a time, starting with the default survey that worked without any changes ...) and eventually you will have a technical detail to share with us (the exact thing you changed last and what stopped working but used to ... along with some CLI output which is handy) or you'll end up with a completely functional system. LOL

4) I like that you said the heck with it and just reinstalled. Very good reflex. Vicibox (thanks Kumba!) is very easy to install, but it's not "simple", so doing it a couple times can make you more familiar with the system and less fearful of changes.

5) We charge $99 for a single campaign setup (even if it's a Survey) on a known working system. For known working: we require it have the ability to work in Autodial (ratio) mode and have sound in both directions when calling (for instance) your cell phone. So your agent phone can hear the cell phone and the cell phone can hear your agent when calling on a Campaign because the agent pressed resume and the system autodialed the cell phone. If that's true, a $99 campaign set fee and *poof* that campaign is a Survey campaign with the proper settings. I realize you did this once and had it working .. but if you want to have another set of eyes do it, that's been our campaign setup fee for quite some time.

6) Beware: I've had clients whose carriers "detected" a survey campaign and turned off DTMF. Be sure you have more than one carrier (even if the other carriers are only for testing ... very helpful in situations like this!)

[Kinghtflyer]

From Iso Vicidial Redux 4.01 VERSION: 2.6-357c BUILD: 121116-1407 Asterisk 1.44-vici Single Server Express Install

I reloaded from iso again and was successful in the tutorials up to survey press-1 type campaign.
But cannot seem to get audio to play.
I am interested in your $99 deal but cannot get phones 2 work both ways in ratio. Manual dial no problem.

[williamconley]

"cannot get phones 2 work both ways in ratio" if this means you get one way audio, you likely have a firewall issue.

When you attempt your first press-1 survey, do not change anything. use the existing setup as it is and see if the sample campaign works. if not ... once again it is possible you have a firewall issue. among other possibilities. consider posting your asterisk command line output from a single attempt (not 3000 lines of unrelated code, of course, just from that single attempt).

[Kinghtflyer]

I can make calls from extension. I don't think it is a firewall issue.

The following is a CLI transcript from my last attempt starting from when I the agent was connected to the conference.



[Nov 20 14:24:23] > Channel SIP/202-00000017 was answered.
[Nov 20 14:24:23] -- Executing [8600051@default:1] MeetMe("SIP/202-00000017", "8600051|F") in new stack
[Nov 20 14:24:23] == Parsing '/etc/asterisk/meetme.conf': [Nov 20 14:24:23] Found
[Nov 20 14:24:23] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Nov 20 14:24:23] Found
[Nov 20 14:24:23] -- Created MeetMe conference 1023 for conference '8600051'
[Nov 20 14:24:23] -- <SIP/202-00000017> Playing 'conf-onlyperson' (language 'en')
[Nov 20 14:24:24] NOTICE[28319]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.1.26
[Nov 20 14:24:25] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 20 14:24:46] == Parsing '/etc/asterisk/manager.conf': [Nov 20 14:24:46] Found
[Nov 20 14:24:46] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 20 14:24:46] -- Executing [915127626397@default:1] AGI("Local/915127626397@default-3a20,2", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 20 14:24:46] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 20 14:24:46] -- Executing [915127626397@default:2] Dial("Local/915127626397@default-3a20,2", "SIP/15127626397@did.voip.les.net||tTo") in new stack
[Nov 20 14:24:46] -- Called 15127626397@did.voip.les.net
[Nov 20 14:24:50] -- SIP/did.voip.les.net-00000018 is ringing
[Nov 20 14:24:50] -- SIP/did.voip.les.net-00000018 is making progress passing it to Local/915127626397@default-3a20,2
[Nov 20 14:25:00] -- SIP/did.voip.les.net-00000018 answered Local/915127626397@default-3a20,2
[Nov 20 14:25:00] > Channel Local/915127626397@default-3a20,1 was answered.
[Nov 20 14:25:00] -- Executing [8366@default:1] Playback("Local/915127626397@default-3a20,1", "sip-silence") in new stack
[Nov 20 14:25:00] -- <Local/915127626397@default-3a20,1> Playing 'sip-silence' (language 'en')
[Nov 20 14:25:00] WARNING[28373]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Nov 20 14:25:00] -- Executing [8366@default:2] AGI("Local/915127626397@default-3a20,1", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 20 14:25:00] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 20 14:25:00] -- Executing [8366@default:3] AGI("Local/915127626397@default-3a20,1", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
[Nov 20 14:25:00] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Nov 20 14:25:02] == Parsing '/etc/asterisk/manager.conf': [Nov 20 14:25:02] Found
[Nov 20 14:25:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 20 14:25:02] == Parsing '/etc/asterisk/manager.conf': [Nov 20 14:25:02] Found
[Nov 20 14:25:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 20 14:25:02] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 20 14:25:02] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Nov 20 14:25:02] -- Executing [8366@default:4] AGI("Local/915127626397@default-3a20,1", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
[Nov 20 14:25:02] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Nov 20 14:25:02] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 20 14:25:02] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 20 14:25:03] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Nov 20 14:25:03] -- Executing [8366@default:5] Hangup("Local/915127626397@default-3a20,1", "") in new stack
[Nov 20 14:25:03] == Spawn extension (default, 8366, 5) exited non-zero on 'Local/915127626397@default-3a20,1'
[Nov 20 14:25:03] -- Executing [h@default:1] DeadAGI("Local/915127626397@default-3a20,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Nov 20 14:25:04] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Nov 20 14:25:04] -- Executing [h@default:1] DeadAGI("Local/915127626397@default-3a20,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----18-----4") in new stack
[Nov 20 14:25:05] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --18-----4 completed, returning 0
[Nov 20 14:25:05] == Spawn extension (default, 915127626397, 2) exited non-zero on 'Local/915127626397@default-3a20,2'
[Nov 20 14:25:07] == Parsing '/etc/asterisk/manager.conf': [Nov 20 14:25:07] Found
[Nov 20 14:25:07] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 20 14:25:07] == Manager 'sendcron' logged off from 127.0.0.1
Vicibox1*CLI>

[williamconley]

Have you tried this in autodial mode before "survey" mode? (as in: following the tutorial in the manual from the beginning ... instead of skipping to the end ...). It is entirely possible you are trying to make a survey campaign work with a dialer that is not fully functional (Darth will tell you that these death stars are still vulnerable if they are not yet fully functional).

[Kinghtflyer]

I must confess, This last go round I followed the tutorial, and skipped over the "setup inward groups. and point the did.
I will return to those steps and rework my way past predictive dialing again. I am pretty sure I witnessed predictive dialing working before attempting the survey.
I will be patient.

[williamconley]

While I agree that neither of those is likely to be necessary, the main goal is to get autodialing working first. Some other day we will discuss your use of the term "predictive" LOL but first get the dialer to dial with a logged in agent in ratio mode (1:1 is fine). After that, it is a simple (?!) ... step-by-step process to switch to "Survey" from "Autodial" (only true if Autodial works!).