Answered calls not being sent to an agent, but get NA status

[MarcoDisco]

Hi,

I have a 3-server vicibox cluster (DB & Web, Archive & Asterisk), created using Vicibox 3.1.14 64-bit.

Astguiclient 2.4-351a
Build 111223-0043
Asterisk: 1.4.38-vici

I'm running an outbound-only campaign. I also have the 'old' hosted vicidial server, which this new cluster is meant to replace.

When compared with the 'old' hosted Vicidial system, it was noticed that agents were having to wait from 1-3 minutes for each call (as opposed to 20-30 seconds on the hosted system).

Also around 90% of outbound calls placed were being logged with the status NA.

When I watch the Asterisk log being generated, I see that many calls placed with our outbound SIP provider show as being answered, but then get hung-up very quickly and are logged with status NA, and are not sent to agents. There are free agents waiting.

Here's an example from the log:

Code: Select all

     -- Executing [90191548nnnn@default:1] AGI("Local/90191548nnnn@default-b57c,2", "agi://127.0.0.1:4577/call_log") in new stack
     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
     -- Executing [90191548nnnn@default:2] Dial("Local/90191548nnnn@default-b57c,2", "SIP/VoipProvider/0191548nnnn||tTor") in new stack
     -- Called VoipProvider/0191548nnnn
     -- SIP/VoipProvider-00000219 is making progress passing it to Local/90191548nnnn@default-b57c,2
     -- SIP/VoipProvider-00000219 answered Local/90191548nnnn@default-b57c,2
        > Channel Local/90191548nnnn@default-b57c,1 was answered.
     -- Executing [8368@default:1] Playback("Local/90191548nnnn@default-b57c,1", "sip-silence") in new stack
     -- <Local/90191548nnnn@default-b57c,1> Playing 'sip-silence' (language 'en')
 file.c: Unexpected control subclass '-1'
     -- Executing [8368@default:2] AGI("Local/90191548nnnn@default-b57c,1", "agi://127.0.0.1:4577/call_log") in new stack
     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
     -- Executing [8368@default:3] AGI("Local/90191548nnnn@default-b57c,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
     -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
     -- Executing [8368@default:4] AGI("Local/90191548nnnn@default-b57c,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
     -- Executing [8368@default:5] Hangup("Local/90191548nnnn@default-b57c,1", "") in new stack
   == Spawn extension (default, 8368, 5) exited non-zero on 'Local/90191548nnnn@default-b57c,1'
     -- Executing [h@default:1] DeadAGI("Local/90191548nnnn@default-b57c,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
     -- Executing [h@default:1] DeadAGI("Local/90191548nnnn@default-b57c,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----2") in new stack
     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----2 completed, returning 0
   == Spawn extension (default, 90191548nnnn, 2) exited non-zero on 'Local/90191548nnnn@default-b57c,2'


I have anonymized the dialled number to 0191548nnnn.

The call is dialled with the VoIP provider, then answered. Next, agi-VDAD_ALL_outbound.agi is called. After the call to agi-VDAD_ALL_outbound.agi, the call gets a hangup() within 2 seconds.

Could anyone suggest why these answered calls are being hung up and not sent to an agent?

Thanks,
Mark

[mflorell]

Looks like the common Local/ channel resolution issue. It is usually caused by one-way audio or no-audio calls, which are usually caused by improper firewall settings or a bad carrier.

[MarcoDisco]

Hi Matt,

Thanks for getting back to me.

Yes, RTP problem. One-sided calls. You are spot on.

I discovered this morning that there was an inconsistency in the RTP ranges between rtp.conf and the rtp range on the firewall. Some calls were negotiating passable rtp ports and being connected, whereas the majority had 'bad' rtp ports so ended up with one-sided calls and were dropped by Vicidial.

All working now and very happy.

Thanks for your input.

Kind Regards,
Mark

[raz]

Hi! I am having the same issue regarding the RTPs you have mentioned. Can you please help me to resolve the issue? Can you please give a step by step procedure on how i could go about in resolving the problem?

[williamconley]

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

[rorstel]

Hi all,

Im really new on this good project, and Im so proud to have access here. So many thanks to give me such opportunity.
Im ready to go with my following system, and have found the same issue posted here.

Not sure if this has been resolved yet, but at least I would like to collaborate finding a solution (at least temporal).
The thing is that Im not relaly sure why asterisk has this behavior:

Vicidial (sip registration, gsm codec) --> (gsm codec) Asterisk PBX (gsm codec) --> (gsm codec) VOIP provider.

Each attempt to call is answered, but it is not being delivered to a free agent. Seems that all is being swallowed by VDAD_ALL_outbound agi.
In logs we have (I have replaced some XXXX in the number dialed):

[Apr 22 03:56:20] -- Executing [6004371075034XXXX@default:1] AGI("Local/6004371075034XXXX@default-00000134;2", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 22 03:56:20] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=QBE))
[Apr 22 03:56:20] -- <Local/6004371075034XXXX@default-00000134;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 22 03:56:20] -- Executing [600437107503414712527@default:2] Dial("Local/6004371075034XXXX@default-00000134;2", "SIP/ABC/03414712527,,tTo") in new stack
[Apr 22 03:56:20] == Using SIP RTP CoS mark 5
[Apr 22 03:56:20] -- Called SIP/ABC/03414712527
[Apr 22 03:56:21] -- SIP/ABC-00000123 is ringing
[Apr 22 03:56:21] -- SIP/ABC-00000123 is making progress passing it to Local/6004371075034XXXX@default-00000134;2
[Apr 22 03:56:48] -- SIP/ABC-00000123 answered Local/6004371075034XXXX@default-00000134;2
[Apr 22 03:56:48] > Channel Local/6004371075034XXXX@default-00000134;1 was answered.
[Apr 22 03:56:48] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 22 03:56:48] -- Executing [8368@default:1] Playback("Local/6004371075034XXXX@default-00000134;1", "sip-silence") in new stack
[Apr 22 03:56:48] -- <Local/6004371075034XXXX@default-00000134;1> Playing 'sip-silence.gsm' (language 'en')
[Apr 22 03:56:48] -- Executing [8368@default:2] AGI("Local/6004371075034XXXX@default-00000134;1", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 22 03:56:48] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=QBE))
[Apr 22 03:56:48] -- <Local/6004371075034XXXX@default-00000134;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 22 03:56:48] -- Executing [8368@default:3] AGI("Local/6004371075034XXXX@default-00000134;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Apr 22 03:56:48] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Apr 22 03:56:49] -- <Local/6004371075034XXXX@default-00000134;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Apr 22 03:56:49] -- Executing [8368@default:4] AGI("Local/6004371075034XXXX@default-00000134;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Apr 22 03:56:49] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Apr 22 03:56:50] -- <Local/6004371075034XXXX@default-00000134;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Apr 22 03:56:50] -- Executing [8368@default:5] Hangup("Local/6004371075034XXXX@default-00000134;1", "") in new stack
[Apr 22 03:56:50] == Spawn extension (default, 8368, 5) exited non-zero on 'Local/6004371075034XXXX@default-00000134;1'
[Apr 22 03:56:50] -- Executing [h@default:1] AGI("Local/60043710750341XXXX@default-00000134;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 22 03:56:51] -- <Local/6004371075034XXXX@default-00000134;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 22 03:56:51] -- Executing [h@default:1] AGI("Local/6004371075034XXXX@default-00000134;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----31-----3") in new stack
[Apr 22 03:56:52] -- <Local/6004371075034XXXX@default-00000134;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --31-----3 completed, returning 0
[Apr 22 03:56:52] == Spawn extension (default, 6004371075034XXXX, 2) exited non-zero on 'Local/6004371075034XXXX@default-00000134;2'

The thing is that with just replacing sip-silence in 8368 context then calls started working properly, being delivered once attended.
Not sure why changing the file in gsm format to the wav format all is good. I tried to look for more updated threads but no success:

Goautodial 3.3 CE
Vicidial VERSION: 2.7-401a
BUILD: 130508-2256

Thanks,