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Ringgroup never transfered to voicemail after timeout

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Cedric Malitte

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Hello,
I'm testing a patton 4112 with 4 FXOs.
Inbound calls are ok, outbound calls are ok.

BUT, if I send the FXOs to a ring group (801), and that this ringroup as a timeout to send the call to an extension voicemail (200), there's a message on the calling phone telling that the voicemail of the extension has not been set.
In the call log I can see the the ringgroup being called, and timing out after 20 seconds, and the call being transfered to VM(200).

Am I missing something ?
 
Last edited:
And if you dial extension 200, and it goes to the VM, what do you get?
 
leejor said:
And if you dial extension 200, and it goes to the VM, what do you get?
Click to expand...
Works fine.

If I call 200, it goes to voicemail of 200 after timeout.
 
Check the 3CX Activity Log to be sure that the call from the ring group is going to the same destination. You should get the same results for both calls.
 
Capture.PNG
This is what I have in the call log.
So it's directed to VM of extension 200 but plays like if the VM was not setup.
But VM is working fine if I call it from another extension, or if I redirect the FXO directly to the voicemail.
 
I would check the Activity Log. It shows the call flow, and why a call goes where it does.
 
When the caller inbound hears this message - do you see they are still connected to 3cx in the current call list? Or are they being redirected to the TelCo ?
 
If I unplug all the phones in the ringroup, the call is directed to the voicemail.
If I leave only one phone and I reject the call, it's sent to voicemail.
Else, if there's a timeout, it's the same message telling the voicemail is not set.

Here's the log
Code: 
05/29/2018 4:23:02 PM - Leg L:12.3[VMail:999] is terminated: Cause: BYE from local
05/29/2018 4:23:02 PM - [CM503008]: Call(C:12): Call is terminated
05/29/2018 4:23:02 PM - Leg L:12.1[Line:10000<<xxxxxxxxxx] is terminated: Cause: BYE from 192.168.20.6:5060
05/29/2018 4:23:01 PM - Currently active calls - 1: [12]
05/29/2018 4:23:00 PM - [CM503007]: Call(C:12): VMail:999 has joined, contact <sip:[email protected]:5483>
05/29/2018 4:23:00 PM - [CM503007]: Call(C:12): Line:10000<<xxxxxxxxxxhas joined, contact <sip:[email protected]:5060>
05/29/2018 4:23:00 PM - L:12.3[VMail:999] has joined to L:12.1[Line:10000<<xxxxxxxxxx]
05/29/2018 4:23:00 PM - [CM505001]: Endpoint VMail:999: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] PBX contact: [sip:[email protected]:5060]
05/29/2018 4:23:00 PM - [CM503002]: Call(C:12): Alerting VMail:999 by contact <sip:[email protected]:5483>
05/29/2018 4:23:00 PM - Leg L:12.2[Extn:202] is terminated: Cause: 487 Request Terminated/INVITE from 127.0.0.1:5488
05/29/2018 4:23:00 PM - [CM503003]: Call(C:12): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5488
05/29/2018 4:23:00 PM - [CM503025]: Call(C:12): Calling T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=f91976ea19b2c43f] for L:12.1[Line:10000<<xxxxxxxxxx]
05/29/2018 4:23:00 PM - [CM503005]: Call(C:12): Forwarding: T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=f91976ea19b2c43f]
05/29/2018 4:23:00 PM - L:12.1[Line:10000<<xxxxxxxxxx] forwards call from RingAll:801 to VMail:999 based on rule
 
Last edited:
So I took a SIP line and directed it o the same ring group, works fine.
I can reject, let timeout the call and it's redirected to the voicemail of ext 200.
Must be something in the Patton config
 
craigreilly said:
When the caller inbound hears this message - do you see they are still connected to 3cx in the current call list? Or are they being redirected to the TelCo ?
Click to expand...
I checked on the patton, and when I have the message, the line seems to be disconnected.
 
It's Bell !
They put a timeout on the line and it's their message !
I will call them to have this removed.

Thanks @craigreilly for pointing me in the direction !
 
Last edited:
Glad to see the issue has been discovered. Let us know if the issue has been resolved.
 
YiannisH_3CX said:
Glad to see the issue has been discovered. Let us know if the issue has been resolved.
Click to expand...
I should be able to verify this on friday as Bell told me it could take between 24-48Hrs for the removal of the voicemail on their side. Looks like they have to manually patch it with wires :)
 
So.... Bell decided that to remove the mail they had to destroy the line.....
I had no line for 72Hrs until a technician came, called his office to re-enable the line.
But, now when I call from outside, the call seems to be routed to VM but on the caller side, it's still ringing and in 3CX the call is dropped.
I do not understand what is causing that.
20.5 is the 3CX, 20.6 is the patton

Code: 
06/02/2018 1:04:19 PM - Leg L:55.2[VMail:999] is terminated: Cause: BYE from local
06/02/2018 1:04:19 PM - [CM503008]: Call(C:55): Call is terminated
06/02/2018 1:04:19 PM - Leg L:55.1[Line:10000<<xxxxxxxxxx] is terminated: Cause: BYE from 192.168.20.6:5060
06/02/2018 1:04:16 PM - [CM503007]: Call(C:55): VMail:999 has joined, contact <sip:[email protected]:5483>
06/02/2018 1:04:16 PM - [CM503007]: Call(C:55): Line:10000<<xxxxxxxxxx has joined, contact <sip:[email protected]:5060>
06/02/2018 1:04:16 PM - L:55.2[VMail:999] has joined to L:55.1[Line:10000<<xxxxxxxxxx]
06/02/2018 1:04:16 PM - [CM505001]: Endpoint VMail:999: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] PBX contact: [sip:[email protected]:5060]
06/02/2018 1:04:16 PM - [CM503002]: Call(C:55): Alerting VMail:999 by contact <sip:[email protected]:5483>
06/02/2018 1:04:16 PM - [CM503025]: Call(C:55): Calling T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=3c9adaea41f15d4d] for L:55.1[Line:10000<<xxxxxxxxxx]
06/02/2018 1:04:16 PM - [CM503027]: Call(C:55): From: Line:10000<<xxxxxxxxxx ("xxxxxx:TP" <sip:[email protected]:5060>)  to  T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=3c9adaea41f15d4d]
06/02/2018 1:04:16 PM - [CM503004]: Call(C:55): Route 1: from L:55.1[Line:10000<<xxxxxxxxxx] to T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=3c9adaea41f15d4d]
06/02/2018 1:04:16 PM - [Flow] Call(C:55): has built target endpoint: VMail:999 for call from L:55.1[Line:10000<<xxxxxxxxxx]
06/02/2018 1:04:16 PM - [Flow] Target endpoint for 200 is VMail:999
06/02/2018 1:04:16 PM - [CM503010]: Call(C:55): Making route(s) from Line:10000<<xxxxxxxxxx to <sip:[email protected]:5060>
06/02/2018 1:04:16 PM - [CM505002]: Gateway:[Entree TP] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Patton SN4114 JO EUI 00A0BA0B461E R6.10 2018-03-09 H323 SIP FXS FXO M5T SIP Stack/4.2.14.18] PBX contact: [sip:[email protected]:5060]
06/02/2018 1:04:16 PM - [CM500002]: Call(C:55): Info on incoming INVITE from Line:10000<<xxxxxxxxxx:
Invite-IN Recv Req INVITE from 192.168.20.6:5060 tid=2d9d242ac87035da5 Call-ID=62eb6140f30d49e4:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.6:5060;branch=z9hG4bK2d9d242ac87035da5
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=udp>
To: <sip:[email protected]:5060>
From: "xxxxxx"<sip:[email protected]:5060>;tag=eb84c6bd2c
Call-ID: 62eb6140f30d49e4
CSeq: 23011 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, INFO, UPDATE, REFER, REGISTER
Content-Type: application/sdp
Proxy-Authorization: Digest username="10000",realm="3CXPhoneSystem",nonce="414d53595b12ce1076:caf8a291376e8722cfea7ef48d39aae5",uri="sip:[email protected]:5060",response="762a2a74581ddd48d13fcb5af597a30f",algorithm=MD5
Supported: replaces
User-Agent: Patton SN4114 JO EUI 00A0BA0B461E R6.10 2018-03-09 H323 SIP FXS FXO M5T SIP Stack/4.2.14.18
Content-Length: 244

v=0
o=MxSIP 0 4 IN IP4 192.168.20.6
s=SIP Call
c=IN IP4 192.168.20.6
t=0 0
m=audio 4866 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
06/02/2018 1:04:16 PM - [CM503001]: Call(C:55): Incoming call from Line:10000<<xxxxxxxxxx to <sip:[email protected]:5060>
06/02/2018 1:04:16 PM - Line limit check: Current # of calls for line Lc:10000(@Entree TP[<sip:[email protected]:5060>]) is 1; limit is 4
06/02/2018 1:04:16 PM - [CM503012]: Inbound out-of-office hours rule (TP) for 10000 forwards to VM:200
06/02/2018 1:04:16 PM - [Flow] Looking for inbound target: called=000000; caller="xxxxxx" <sip:xxxxxxxxxx@:5060>
 
Taking the VM option off the line should have been a matter of removing two call forward options (call forward busy and call forward no answer) from your line. Sounds like someone misunderstood and thought it was all to come out.

On an incoming call, is the extension (or any other extension you route it to) able to answer the call before it goes to voicemail? Are you able to call out on the line?
 
Last edited:
leejor said:
Taking the VM option off the line should have been a matter of removing two fall forward options (call forward busy and call forward no answer) from your line. Sounds like someone misunderstood and thought it was all to come out.

On an incoming call, is the extension (or any other extension you route it to) able to answer the call before it goes to voicemail? Are you able to call out on the line?
Click to expand...

I took the setup home to test it further this week end.
The call is directed to the voicemail because of the schedule.
Incoming route is a ring group during business hours and voicemail of an extension(200) the rest of the time.

I will try another extension and without voicemail.
 
I guess I have a trouble in between the Patton and 3CX.
If I setup the 000000 DID to be routed to an extension, the extension rings, and when I pickup the call, 2 seconds later the phone hangs up and begins to ring again.
On the caller side, it's still ringing.
I tested a SIP trunk, no trouble.
 
It sounds as if messages are not getting back to the Patton, but, it may also be a configuration issue. I assume that all devices are local. That the gateway was provisioned using a file from 3CX.

Do outgoing calls seem to have any issues?

You have confirmed that your 3CX trunk group settings (in particular the IP and port) are correct?

If you are about to pull out your hair, do a full reset on the gateway, be sure it has the latest approved firmware, and re-configure.

Have you tried plugging an analogue set into the phone jack (unplugging the Patton) , try an incoming call just to confirm that the phone line, is, in fact, working properly.
 
I tried inbound and outbound calls => nothing passed in or out.
I checked the debug in Patton, and looks like it's dropping the line but can't figure the reason.

Phone line works with a phone plugged to it.
I changed my phone cable, just in case: not better, not worse. Still not working.

I did reset the patton to factory configuration and reloaded the generated cfg file.

Config file:

Code: 
#########################################################
#
# For Patton 2 to 8 FXO
# Supported firmware versions R6.x
#
# Author: Tommaso Missoni
# Contact: [email protected]
#
#########################################################

webserver port 80 language en

system

  ic voice 0
    low-bitrate-codec g729
    
sntp-client

sntp-client server primary pool.ntp.org

profile ppp default


profile call-progress-tone defaultDialtone
  flush-play-list
  play 1 1000 350 -13 440 -13

profile call-progress-tone defaultAlertingtone
  flush-play-list
  play 1 1000 440 -19 480 -19
  pause 2 3000

profile call-progress-tone defaultBusytone
  flush-play-list
  play 1 500 480 -24 620 -24
  pause 2 500

profile call-progress-tone defaultReleasetone
  flush-play-list
  play 1 250 480 -24 620 -24
  pause 2 250

profile call-progress-tone defaultCongestiontone
  flush-play-list
  play 1 250 480 -24 620 -24
  pause 2 250
        

profile tone-set default

profile voip default
  codec 1 g711ulaw64k rx-length 20 tx-length 20 silence-suppression
  codec 2 g711alaw64k rx-length 20 tx-length 20 silence-suppression
  codec 3 g729 rx-length 20 tx-length 20 silence-suppression
  fax transmission 1 relay t38-udp

profile pstn default
  echo-canceler-hybrid-loss 0
  echo-canceler-nlp silence

profile sip default
 
profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_WAN
    ipaddress dhcp
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface IF_IP_LAN
    ipaddress 192.168.20.6 255.255.255.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context cs switch
 
 
 routing-table called-e164 RT_SIP_TO_FXO
    route .% dest-service HG_TO_FXO
    
  routing-table called-e164 RT_FXO_00_TO_SIP   
    route default dest-interface IF_SIP    MT_CALL_ID_00
    
  routing-table called-e164 RT_FXO_01_TO_SIP   
    route default dest-interface IF_SIP MT_CALL_ID_01
    
  routing-table called-e164 RT_FXO_02_TO_SIP   
    route default dest-interface IF_SIP MT_CALL_ID_02
    
  routing-table called-e164 RT_FXO_03_TO_SIP   
    route default dest-interface IF_SIP MT_CALL_ID_03
    
 mapping-table called-e164 to called-e164 MT_CALL_ID_00
    map .% to 000000

  mapping-table called-e164 to called-e164 MT_CALL_ID_01
    map .% to 000001

  mapping-table called-e164 to called-e164 MT_CALL_ID_02
    map .% to 000002

  mapping-table called-e164 to called-e164 MT_CALL_ID_03
    map .% to 000003

    
 service hunt-group HG_TO_FXO   
    
    drop-cause destination-out-of-order
    drop-cause user-busy
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable   
    
        
        route call 1 dest-interface IF_FXO_0
        route call 2 dest-interface IF_FXO_1
        route call 3 dest-interface IF_FXO_2
        route call 4 dest-interface IF_FXO_3
        
 
  interface sip IF_SIP
    bind context sip-gateway GW_SIP
    early-disconnect
    route call dest-table RT_SIP_TO_FXO
    remote 192.168.20.5 5060
    trust remote
    address-translation outgoing-call request-uri user-part fix 10000 host-part to-header target-param none
    address-translation incoming-call called-e164 request-uri

  interface fxo IF_FXO_0
    route call dest-table RT_FXO_00_TO_SIP
    loop-break-duration min 300 max 1200
    caller-id format bell
    ring-number on-caller-id
    mute-dialing
    disconnect-signal loop-break
    disconnect-signal busy-tone
    dial-after dialtone 
    
  interface fxo IF_FXO_1
    route call dest-table RT_FXO_01_TO_SIP
    loop-break-duration min 300 max 1200
    caller-id format bell
    ring-number on-caller-id
    mute-dialing
    disconnect-signal loop-break
    disconnect-signal busy-tone
    dial-after dialtone 
    
    
  interface fxo IF_FXO_2
    route call dest-table RT_FXO_02_TO_SIP
    loop-break-duration min 300 max 1200
    caller-id format bell
    ring-number on-caller-id
    mute-dialing
    disconnect-signal loop-break
    disconnect-signal busy-tone
    dial-after dialtone 
 
  interface fxo IF_FXO_3
route call dest-table RT_FXO_03_TO_SIP
    loop-break-duration min 300 max 1200
    caller-id format bell
    ring-number on-caller-id
    mute-dialing
    disconnect-signal loop-break
    disconnect-signal busy-tone
    dial-after dialtone 

context cs switch
  no shutdown

authentication-service AS_ALL_LINES
  realm 1 3CXPhoneSystem
  username 10000 password xxxxxx

location-service LS_10000
  domain 1 192.168.20.5
 
  identity-group default
    authentication outbound
      authenticate 1 authentication-service AS_ALL_LINES username 10000
 
  identity 10000
    authentication outbound
      authenticate 1 authentication-service AS_ALL_LINES

    registration outbound
      registrar 192.168.20.5 5060
      lifetime 300
      register auto

      register auto
      
context sip-gateway GW_SIP

  interface LAN
    bind interface IF_IP_LAN context router port 5060

context sip-gateway GW_SIP
  bind location-service LS_10000
  no shutdown
 
port ethernet 0 0
  medium auto
  encapsulation ip
  bind interface IF_IP_LAN router
  no shutdown
 
 port ethernet 0 1
  medium auto
  encapsulation ip
  bind interface IF_IP_WAN router
  no shutdown

port fxo 0 0
  use profile fxo us
  encapsulation cc-fxo
  bind interface IF_FXO_0 switch
  no shutdown

port fxo 0 1
  use profile fxo us
  encapsulation cc-fxo
  bind interface IF_FXO_1 switch
  no shutdown
 
port fxo 0 2
  use profile fxo us
  encapsulation cc-fxo
  bind interface IF_FXO_2 switch
  no shutdown

port fxo 0 3
  use profile fxo us
  encapsulation cc-fxo
  bind interface IF_FXO_3 switch
  no shutdown
 
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