Toggle sidebar Toggle sidebar

Call ID desplay Anonymous

Status
Not open for further replies.

Gnomx

Free User
Joined
May 2, 2019
Messages
23
Reaction score
1
Hi guys.
I have a problem with my 3cx software. How can I show on display inbound call? For now i get Anonymous , no number shows on display.

Cheers.
 
Hello @ Gnomx

There are several parameters to consider that affect the inbound caller ID. The first thing you need to check is if your provider is sending you a caller ID in the Invite.
Then you need to find out in what SIP field the caller ID is available. Once you know that, you need to adjust your Inbound parameters to read the callers name and number from the correct SIP fields.
 
  • Like
Reactions: AWS2P
Ok. So how can I know if ISP sends to me called ID , DO I have to call them or there is a tools that can help me on this?
Thanks
 
Perhaps you can start searching info in "activity log" from dashboard and see if caller number appears or if anonymous is present .
just do an incoming call to your pbx from mobile for example and follow what happens in log
 
aws2p said:
Perhaps you can start searching info in "activity log" from dashboard and see if caller number appears or if anonymous is present .
just do an incoming call to your pbx from mobile for example and follow what happens in log
Click to expand...
Hi.
I've tried to call local number here and this is the log :
"16/05/2019 15:18:27 - [CM503003]: Call(C:61): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Cancelled/INVITE from 192.168.1.31:5060 "
No number appears.

Cheers
 
YiannisH_3CX said:
Hello @ Gnomx

There are several parameters to consider that affect the inbound caller ID. The first thing you need to check is if your provider is sending you a caller ID in the Invite.
Then you need to find out in what SIP field the caller ID is available. Once you know that, you need to adjust your Inbound parameters to read the callers name and number from the correct SIP fields.
Click to expand...
I call ISP and they told me that service ID is aviable on this line. We must check Patton or 3cx.

Here a mine patton running config:


Code: 
#----------------------------------------------------------------#

#                                                                #

# SN4114/JO/EUI                                                  #

# R6.9 2017-03-13 H323 SIP FXS FXO                               #

# 1970-01-03T06:08:17                                            #

# SN/00A0BA0DC9BF                                                #

# Generated configuration file                                   #

#                                                                #

#----------------------------------------------------------------#



cli version 3.20

clock local default-offset +00:00

webserver port 80 language en



system



  ic voice 0

    low-bitrate-codec g729



profile ppp default



profile call-progress-tone defaultDialtone

  play 1 200 425 0

  pause 2 200

  play 3 600 425 0

  pause 4 1000



profile call-progress-tone defaultBusytone

  play 1 200 425 -7

  pause 2 200



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

  no autonomous-transitioning



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 192.168.1.4 255.255.255.0

    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



  interface sip IF_SIP

    bind context sip-gateway GW_SIP

    route call dest-table RT_SIP_TO_FXO

    remote 192.168.1.190 5060

    early-disconnect

    address-translation outgoing-call request-uri user-part fix 10000 host-part to-header target-param none

    trust remote



  interface fxo IF_FXO_0

    route call dest-table RT_FXO_00_TO_SIP

    loop-break-duration min 60 max 1000

    disconnect-signal loop-break

    disconnect-signal busy-tone

    ring-number on-caller-id

    dial-after timeout 1

    mute-dialing



  interface fxo IF_FXO_1

    route call dest-table RT_FXO_01_TO_SIP

    loop-break-duration min 60 max 1000

    disconnect-signal loop-break

    disconnect-signal busy-tone

    ring-number on-caller-id

    dial-after timeout 1

    mute-dialing



  interface fxo IF_FXO_2

    route call dest-table RT_FXO_02_TO_SIP

    loop-break-duration min 60 max 1000

    disconnect-signal loop-break

    disconnect-signal busy-tone

    ring-number on-caller-id

    dial-after timeout 1

    mute-dialing



  interface fxo IF_FXO_3

    route call dest-table RT_FXO_03_TO_SIP

    loop-break-duration min 60 max 1000

    disconnect-signal loop-break

    disconnect-signal busy-tone

    ring-number on-caller-id

    dial-after timeout 1

    mute-dialing



  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



context cs switch

  no shutdown



authentication-service AS_ALL_LINES

  realm 1 3CXPhoneSystem

  username 10000 password 8g/s9s6iL+74Pu5YxZuvBQ== encrypted



location-service LS_10000

  domain 1 192.168.1.190



  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 192.168.1.190 5060

      lifetime 300

      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 fxo 0 0

  encapsulation cc-fxo

  bind interface IF_FXO_0 switch

  no shutdown



port fxo 0 1

  encapsulation cc-fxo

  bind interface IF_FXO_1 switch

  no shutdown



port fxo 0 2

  encapsulation cc-fxo

  bind interface IF_FXO_2 switch

  shutdown



port fxo 0 3

  encapsulation cc-fxo

  bind interface IF_FXO_3 switch

  shutdown
 
If that is your trunk (FXO) gateway, using a PSTN connection, the parameters you listed don't show what type of Caller ID has been configured. The type used in the gateway must match that used by your provider.
 
leejor said:
If that is your trunk (FXO) gateway, using a PSTN connection, the parameters you listed don't show what type of Caller ID has been configured. The type used in the gateway must match that used by your provider.
Click to expand...
So What do I have to do?? I'm a bit confused... :(
 
What type of caller ID is used by your provider? You then have to confirm and change to that type on the gateway. there may be other settings that can affect this as well even if the caller ID type is correct. Keep in mind that some providers other than the incumbent phone company, have been known to use a different type than is commonly used in the rest of a country/region.

Have you used a caller ID device (analogue set with a display, or stand alone display box), and plugged it directly into one of the PSTN lines to confirm that caller Id is, in fact, being sent? If that doesn't work, then contact your provider, before making any changes on the gateway.
 
Last edited:
Agree with the above, confirm with the provider first or everything else in this ticket string is a waste of time.

Some providers will charge extra for CLID on the line.

If they can't help I can send you some commands for the CLI of the Patton which could potentially provide the information you need.
 
eddv123 said:
Agree with the above, confirm with the provider first or everything else in this ticket string is a waste of time.

Some providers will charge extra for CLID on the line.

If they can't help I can send you some commands for the CLI of the Patton which could potentially provide the information you need.
Click to expand...
Ok brother. I need you help on this... Please tell me the CLI command in order to retrive info.
 
Commands for the FXO device should be:

enable
debug call-router
debug call-control
debug fxo
debug ccfxo
debug context sip-gateway signaling detail 5
debug context sip-gateway transport detail 5
debug context sip-gateway error

Run these, replicate the issue and send over to me as an attached notepad file (not in the body of the string) I will then take a look, also the CLI that it should be.

You can stop the output of the above with the command "no debug all".
 
eddv123 said:
Commands for the FXO device should be:

enable
debug call-router
debug call-control
debug fxo
debug ccfxo
debug context sip-gateway signaling detail 5
debug context sip-gateway transport detail 5
debug context sip-gateway error

Run these, replicate the issue and send over to me as an attached notepad file (not in the body of the string) I will then take a look, also the CLI that it should be.

You can stop the output of the above with the command "no debug all".
Click to expand...
How can I open terminal in order to using theese command line?
Thnaks.
 
Status
Not open for further replies.

Latest Posts

Forum statistics

Threads
111,924
Messages
589,756
Members
164,798
Latest member
Call_Flow.co.uk
3CX Logo Footer Copyright © Sitemap Privacy
Top Bottom