Inbound DID - Call Drop when Agent Answer

[iboam]

All inbound Call Drop when Agent Answer. Calls route from DID to call menu -> inbound group and gets answered byt the agent but not happen when forwarded to an extension

[williamconley]

Are you saying that the agent's phone rings and they "answer" the phone? Or that the call is dropped in the agent's session and then drops? If these agents are OnHook ... try changing to live agents instead and see if that changes your results.

[iboam]

While Agents are on Pause or the call is waiting for an agent everything is fine, once the agent is active and they get the call, it drops immediately

[williamconley]

"get the call" ... I'll need a description of this. Are the agents OnHook agents (their phone rings, then they "answer", then the call drops)? (IE: Agents using phones with OnHook = YES)

If so: You may find that sip debugging shows a missing codec or other handshake issue.

If not: you may try agi debug and see where the "logic" decides to hang up, which may provide the reason.

[iboam]

i mean when they answer the call ....
i need help ... please ...

[marzo]

Hello,

I have a cluster of Vicidial with:
1. telephony server: Vicibox_v.6.0.x86_64-6.0.3 from .iso | Vicidial 2.12-505a Build150806-134 | Asterisk 1.8.32.3-vici | Sangoma A104E card | No Extra Software After Installation | ProLiant DL360 G7 | Intel(R) Xeon(R) CPU X5650 @ 2.67GHz
2. database server: Vicibox_v.6.0.x86_64-6.0.3 from .iso | Vicidial 2.12-505a Build150806-134 | No Extra Software After Installation | 440BX Desktop Reference Platform | Intel(R) Xeon(R) CPU E5-2680 0 @ 2.70GHz | VMware ESXi 6.0
3. web server: Vicibox_v.6.0.x86_64-6.0.3 from .iso | No Extra Software After Installation | 440BX Desktop Reference Platform | Intel(R) Xeon(R) CPU E5-2680 0 @ 2.70GHz | VMware ESXi 6.0

I have the same problem.
In the Export Call Report:
Status. user. full_name. campaign_id. status_name
DROP. 2133. Juliana Rincon. SAC. Agent Not Available

The questions is why do I have a DROP call and have user at the same time.

Thanks in advanced

[williamconley]

marzo:

asterisk CLI and related screen output (screen -list shows them, screen -r {name} attaches, ctrlA-ctrlD detaches). Probably want to view the autodial screen. From a single instance.
Also: if there was more than one call, try bringing the system down to a single call for the test. if that works, but full production doesn't, you have found the flaw with your cool virtualized system. Can't handle a load.

iboam:

Code: Select all

Executing [83047777777777@vicidial-auto:1] Answer("Local/010*001*020*035*78600052@default-00000606;1", "") in new stack

There was no channel in the answer? Or several other entries? Why? If you're deleting them, change them instead of deleting them. Are these onhook agents? Do the agents hear the "ding"? (which shows before the call ends).

Try sip debug on the IP address of the user in question. You may find a handshake issue.

[iboam]

Yes hear the "ding" and then call drops, i have forwarded all my did's to a landline, all forwarded calls are fine, i left only a test did to solve this issue, is there any settings missing for inbound calls in a cluster ?

[williamconley]

Try sip debug on the IP address of the user in question. You may find a handshake issue.

[iboam]

[Apr 22 15:01:44] <--- SIP read from UDP:54.172.60.2:5060 --->
[Apr 22 15:01:44] ACK sip:+1XXXXXXXXXX@10.1.20.35:5060 SIP/2.0
[Apr 22 15:01:44] Call-ID: 554490ee9adde7b91b147c4f80670948@0.0.0.0
[Apr 22 15:01:44] CSeq: 825798 ACK
[Apr 22 15:01:44] From: "MIAMI FL" <sip:+17863251053@my.pstn.sip-provider>;tag=67703007_6772d868_d560ebac-c6a6-4578-baf0-d255ba3ac8f5
[Apr 22 15:01:44] To: <sip:+1XXXXXXXXXX@10.1.20.35;user=phone>;tag=as71f0f2e8
[Apr 22 15:01:44] Max-Forwards: 69
[Apr 22 15:01:44] User-Agent: SIP-
[Apr 22 15:01:44] X-SIP--CallSid: CA64d9a7b215ae6a5eb13342a68ed2bae1
[Apr 22 15:01:44] Via: SIP/2.0/UDP 54.172.60.2:5060;branch=z9hG4bKbdda.d7d7cf03.2
[Apr 22 15:01:44] Via: SIP/2.0/UDP 172.18.18.10:5060;received=54.172.60.14;branch=z9hG4bKd560ebac-c6a6-4578-baf0-d255ba3ac8f5_6772d868_945203764802184
[Apr 22 15:01:44] Content-Length: 0
[Apr 22 15:01:44]
[Apr 22 15:01:44] <------------->
[Apr 22 15:01:44] --- (11 headers 0 lines) ---
[Apr 22 15:01:44] > 0x7f72719e1720 -- Probation passed - setting RTP source address to 54.172.60.14:17750
[Apr 22 15:01:44] -- Executing [99909*4***DID@default:2] AGI("SIP/SIP-Provider-00000784", "agi-VDAD_ALL_inbound.agi") in new stack
[Apr 22 15:01:44] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Apr 22 15:01:44] -- <SIP/SIP-Provider-00000784> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 22 15:01:44] -- <SIP/SIP-Provider-00000784> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 22 15:01:44] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 22 15:01:44] -- Executing [010*001*020*035*78600052@default:1] Goto("Local/010*001*020*035*78600052@default-00000784;2", "default,78600052,1") in new stack
[Apr 22 15:01:44] -- Goto (default,78600052,1)
[Apr 22 15:01:44] -- Executing [78600052@default:1] MeetMe("Local/010*001*020*035*78600052@default-00000784;2", "8600052,Fq") in new stack
[Apr 22 15:01:44] > Channel Local/010*001*020*035*78600052@default-00000784;1 was answered
[Apr 22 15:01:44] -- Executing [83047777777777@vicidial-auto:1] Answer("Local/010*001*020*035*78600052@default-00000784;1", "") in new stack
[Apr 22 15:01:44] -- Executing [83047777777777@vicidial-auto:2] Playback("Local/010*001*020*035*78600052@default-00000784;1", "ding") in new stack
[Apr 22 15:01:44] -- <Local/010*001*020*035*78600052@default-00000784;1> Playing 'ding.gsm' (language 'en')
[Apr 22 15:01:44] -- Executing [83047777777777@vicidial-auto:3] Hangup("Local/010*001*020*035*78600052@default-00000784;1", "") in new stack
[Apr 22 15:01:44] == Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/010*001*020*035*78600052@default-00000784;1'
[Apr 22 15:01:44] -- Executing [h@vicidial-auto:1] AGI("Local/010*001*020*035*78600052@default-00000784;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 22 15:01:44] -- <Local/010*001*020*035*78600052@default-00000784;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 22 15:01:44] == Spawn extension (default, 78600052, 1) exited non-zero on 'Local/010*001*020*035*78600052@default-00000784;2'
[Apr 22 15:01:44] -- Executing [h@default:1] AGI("Local/010*001*020*035*78600052@default-00000784;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 22 15:01:44] -- <Local/010*001*020*035*78600052@default-00000784;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 22 15:01:45] -- <SIP/SIP-Provider-00000784> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 22 15:01:45] -- <SIP/SIP-Provider-00000784> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 22 15:01:45] -- <SIP/SIP-Provider-00000784>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Apr 22 15:01:45] -- Executing [99909*4***DID@default:3] Hangup("SIP/SIP-Provider-00000784", "") in new stack
[Apr 22 15:01:45] == Spawn extension (default, 99909*4***DID, 3) exited non-zero on 'SIP/SIP-Provider-00000784'
[Apr 22 15:01:45] -- Executing [h@default:1] AGI("SIP/SIP-Provider-00000784", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 22 15:01:45] -- <SIP/SIP-Provider-00000784>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 22 15:01:45] Scheduling destruction of SIP dialog '554490ee9adde7b91b147c4f80670948@0.0.0.0' in 6400 ms (Method: ACK)
[Apr 22 15:01:45] set_destination: Parsing <sip:54.172.60.2:5060;lr;ftag=67703007_6772d868_d560ebac-c6a6-4578-baf0-d255ba3ac8f5> for address/port to send to
[Apr 22 15:01:45] set_destination: set destination to 54.172.60.2:5060
[Apr 22 15:01:45] Reliably Transmitting (NAT) to 54.172.60.2:5060:
[Apr 22 15:01:45] BYE sip:+17863251053@172.18.18.10:5060;transport=udp SIP/2.0
[Apr 22 15:01:45] Via: SIP/2.0/UDP my-nat-ip:5060;branch=z9hG4bK660cd2a4;rport
[Apr 22 15:01:45] Route: <sip:54.172.60.2:5060;lr;ftag=67703007_6772d868_d560ebac-c6a6-4578-baf0-d255ba3ac8f5>
[Apr 22 15:01:45] Max-Forwards: 70
[Apr 22 15:01:45] From: <sip:+1XXXXXXXXXX@10.1.20.35;user=phone>;tag=as71f0f2e8
[Apr 22 15:01:45] To: "MIAMI FL" <sip:+17863251053@my.pstn.sip-provider>;tag=67703007_6772d868_d560ebac-c6a6-4578-baf0-d255ba3ac8f5
[Apr 22 15:01:45] Call-ID: 554490ee9adde7b91b147c4f80670948@0.0.0.0
[Apr 22 15:01:45] CSeq: 102 BYE
[Apr 22 15:01:45] User-Agent: Asterisk PBX 11.21.0-vici
[Apr 22 15:01:45] X-Asterisk-HangupCause: Normal Clearing
[Apr 22 15:01:45] X-Asterisk-HangupCauseCode: 16
[Apr 22 15:01:45] Content-Length: 0
[Apr 22 15:01:45]
[Apr 22 15:01:45]
[Apr 22 15:01:45] ---
[Apr 22 15:01:45]
[Apr 22 15:01:45] <--- SIP read from UDP:54.172.60.2:5060 --->
[Apr 22 15:01:45] SIP/2.0 200 OK
[Apr 22 15:01:45] CSeq: 102 BYE
[Apr 22 15:01:45] Call-ID: 554490ee9adde7b91b147c4f80670948@0.0.0.0
[Apr 22 15:01:45] From: <sip:+1XXXXXXXXXX@10.1.20.35;user=phone>;tag=as71f0f2e8
[Apr 22 15:01:45] To: "MIAMI FL" <sip:+17863251053@my.pstn.sip-provider>;tag=67703007_6772d868_d560ebac-c6a6-4578-baf0-d255ba3ac8f5
[Apr 22 15:01:45] Via: SIP/2.0/UDP 10.1.20.35:5060;received=10.1.20.35;branch=z9hG4bK660cd2a4;rport=5060
[Apr 22 15:01:45] Server: SIP-
[Apr 22 15:01:45] X-SIP--CallSid: CA64d9a7b215ae6a5eb13342a68ed2bae1
[Apr 22 15:01:45] Content-Length: 0
[Apr 22 15:01:45]
[Apr 22 15:01:45] <------------->
[Apr 22 15:01:45] --- (9 headers 0 lines) ---
[Apr 22 15:01:45] SIP Response message for INCOMING dialog BYE arrived
[Apr 22 15:01:45] Really destroying SIP dialog '554490ee9adde7b91b147c4f80670948@0.0.0.0' Method: ACK
[Apr 22 15:01:45] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 22 15:01:46]
[Apr 22 15:01:46] <--- SIP read from UDP:10.1.100.54:5060 --->
[Apr 22 15:01:46]
[Apr 22 15:01:46]
[Apr 22 15:01:46] <------------->
[Apr 22 15:01:50]
[Apr 22 15:01:50] <--- SIP read from UDP:10.1.101.84:5060 --->
[Apr 22 15:01:50]
[Apr 22 15:01:50]
[Apr 22 15:01:50] <------------->

[williamconley]

[Apr 22 15:01:45] BYE sip:+17863251053@172.18.18.10:5060;transport=udp SIP/2.0

This appears to be the trigger moment. But I don't read these often enough to know if this originated from your server or the called party or the calling party. Usually if there's a handshake issue there's a clear message like to translation path available for codec g729.

That being said: Are you using g729? I don't see the handshake information where that happened (which is at the start of the call, as a rule).

You could also try turning on agi debugging, in case the cause is a vicidial agi script (agi debugging would share the details of the decision). You can also try to dig through the vicidial logs for reasoning behind any decisions that were made during the routing of this call (var/log/astguiclient).

[iboam]

no, im not using g729, i just install Cluster Server 1 DB / 1 WEB / 1 ASTX, then on the web server i just copy paste settings and procedures as the first i installed before as vicidial-express ...

[williamconley]

oOOOOOoo

Vicibox express ... how did you Cluster these servers?

Vicibox STANDARD will cluster for you. Express does not, last time I checked.

Have you tried this without the cluster? Single standalone server?

[iboam]

I did a express installation for a single server before and it was running fine but now i have i have a cluster server 1 DB / 1 WEB / 1 ASTX, i mean copy paste for (carrier, did, inbound ...)

[iboam]

There is any particular settings for inbound calls using a cluster ????

this is my trunkinbound @ extensions.conf

Code: Select all

[trunkinbound]
; DID call routing process
; use this one instead of the one below if you are having delay issues, and match to number of received digits
; exten => _XXXXXXXXXX,n,AGI(agi-DID_route.agi)

; For even more speed, you can put in the real DIDs instead of XXXXXXXXXX (remove the _ in that case)
exten => _X.,1,NoOp(X,${EXTEN},${CALLERID(NUM)})
exten => _X.,n,AGI(agi-DID_route.agi)

; Catch all even with unexpected first digits (will show in CLI for troubleshooting)
exten => _!,1,Noop(!,${EXTEN},${CALLERID(NUM)})
exten => _!,n,AGI(agi-DID_route.agi)
exten => s,1,NoOp(${EXTEN},${CALLERID(NUM)})
exten => s,n,AGI(agi-DID_route.agi)

; removal of Leading Plus Sign from the the phone
exten => _+X!,1,Noop(Stripping + from start of number, for annoying carriers who insist)
exten => _+X!,n,Goto(trunkinbound,${EXTEN:1},1)

; removal of Leading 1 from the the phone (US numbers only, for annoying carriers who insist)
;exten => _1!,1,Noop(Stripping 1 from start of number)
;exten => _1!,n,Goto(trunkinbound,${EXTEN:1},1)

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


[williamconley]

There is any particular settings for inbound calls using a cluster ????

The only requirement is that the carrier be installed on all dialer/carrier/trunk servers (ie: Not needed on servers that have ONLY agents or only web/db, but any server that will communicate with the carrier for inbound must have the admin->carrier assignment. In theory there is now an "All" option which is very handy for carrier assignment (haven't tested this myself yet).