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Open port 5060 udp

moman62
Level 1
Level 1

Hello, I have a cisco 2921 router and I would like to know how to enable port 5060 so that my ITSP can send information so I can perform outbound/inbound calls.

43 Replies 43

Can you please post the configuration you have for dial peers in your gateway?

 



Response Signature


Here is my ccsip messages and dial-peers;

 

 

I have been following this thread and it's obvious that the guys here are doing there best to help you but they can't work with what you are providing. You do not have the skill set to handle this task and might be best for you to delegate this to someone else who can take this much further. There has been a lot of back and forth and I will try to summarize what the guys have been trying to tell you.

1. You need network connectivity to your ITSP to setup CUBE integration. What network connectivity are you using? MPLS? Dedicated edge router connected to your CUBE? Internet based ITSP?

2. What is the topology of your setup? Can you draw a schematic of how your CUBE connects to your ITSP?

3. Is ICMP enabled on your ITSP signalling IP..Do you know what the Signalling IP is? If it is, can you ping it?

 

Until you can answer these questions, don't try and debug or do any test calls it might just amount to a waste of time and effort...

Please rate all useful posts

dp2 from 2020-04-07 09-42-05.png

Your bind statements looks inaccurate. You have different interfaces on the inbound and outbound dial-peers in each direction. Normally you would have the one interface on the inbound and outbound dial peers to your ITSP and another towards your phone system.



Response Signature


Here is my ccapi:

 

2921LAB#u allterm mondebug voip ccapi inout
voip ccapi inout debugging is on
2921LAB#term mon
2921LAB#
2921LAB#
2921LAB#
2921LAB#
2921LAB#
*Apr 9 20:32:07.757: //-1/0423E2C9862A/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=2135689903
----- ccCallInfo IE subfields -----
cisco-ani=2135689903
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=9093303621
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0

*Apr 9 20:32:07.757: //-1/0423E2C9862A/CCAPI/cc_api_call_setup_ind_common:
Interface=0x224B0288, Call Info(
Calling Number=2135689903,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=9093303621(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=101, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=17597
*Apr 9 20:32:07.757: //-1/0423E2C9862A/CCAPI/ccCheckClipClir:
In: Calling Number=2135689903(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Apr 9 20:32:07.757: //-1/0423E2C9862A/CCAPI/ccCheckClipClir:
Out: Calling Number=2135689903(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Apr 9 20:32:07.757: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Apr 9 20:32:07.757: :cc_get_feature_vsa malloc success
*Apr 9 20:32:07.757: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Apr 9 20:32:07.757: cc_get_feature_vsa count is 1
*Apr 9 20:32:07.757: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Apr 9 20:32:07.757: :FEATURE_VSA attributes are: feature_name:0,feature_time:3287725536,feature_id:92
*Apr 9 20:32:07.757: //17597/0423E2C9862A/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=2135689903(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=9093303621(TON=Unknown, NPI=Unknown))
*Apr 9 20:32:07.761: //17597/0423E2C9862A/CCAPI/cc_process_call_setup_ind:
Event=0x22A64500
*Apr 9 20:32:07.761: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 9093303621
*Apr 9 20:32:07.761: //17597/0423E2C9862A/CCAPI/ccCallSetContext:
Context=0x3FD6F658
*Apr 9 20:32:07.761: //17597/0423E2C9862A/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 17597 with tag 101 to app "_ManagedAppProcess_Default"
2921LAB#
*Apr 9 20:32:07.761: //17597/0423E2C9862A/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
*Apr 9 20:32:07.761: //17597/0423E2C9862A/CCAPI/ccCallDisconnect:
Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Apr 9 20:32:07.761: //17597/0423E2C9862A/CCAPI/ccCallDisconnect:
Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
2921LAB#
*Apr 9 20:32:14.257: //-1/0803B60C8630/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=2135689903
----- ccCallInfo IE subfields -----
cisco-ani=2135689903
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=9093303621
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0

*Apr 9 20:32:14.257: //-1/0803B60C8630/CCAPI/cc_api_call_setup_ind_common:
Interface=0x224B0288, Call Info(
Calling Number=2135689903,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=9093303621(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=101, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=17599
*Apr 9 20:32:14.257: //-1/0803B60C8630/CCAPI/ccCheckClipClir:
In: Calling Number=2135689903(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Apr 9 20:32:14.257: //-1/0803B60C8630/CCAPI/ccCheckClipClir:
Out: Calling Number=2135689903(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Apr 9 20:32:14.257: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Apr 9 20:32:14.257: :cc_get_feature_vsa malloc success
*Apr 9 20:32:14.257: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Apr 9 20:32:14.257: cc_get_feature_vsa count is 2
*Apr 9 20:32:14.257: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Apr 9 20:32:14.257: :FEATURE_VSA attributes are: feature_name:0,feature_time:3287725760,feature_id:93
*Apr 9 20:32:14.257: //17599/0803B60C8630/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=2135689903(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=9093303621(TON=Unknown, NPI=Unknown))
*Apr 9 20:32:14.261: //17599/0803B60C8630/CCAPI/cc_process_call_setup_ind:
Event=0x22A64500
*Apr 9 20:32:14.261: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 9093303621
*Apr 9 20:32:14.261: //17599/0803B60C8630/CCAPI/ccCallSetContext:
Context=0x3FD70C10
*Apr 9 20:32:14.261: //17599/0803B60C8630/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 17599 with tag 101 to app "_ManagedAppProcess_Default"
2921LAB#
*Apr 9 20:32:14.261: //17599/0803B60C8630/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
*Apr 9 20:32:14.261: //17599/0803B60C8630/CCAPI/ccCallDisconnect:
Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Apr 9 20:32:14.261: //17599/0803B60C8630/CCAPI/ccCallDisconnect:
Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
2921LAB#u all\

From what I can tell by looking at the output on my mobile device it would appear that there is no outbound dial peer matched.



Response Signature



@Roger Kallberg wrote:

From what I can tell by looking at the output on my mobile device it would appear that there is no outbound dial peer matched.


OP - you dialled "909330362".  Which dial peer is that supposed to match?

Have you resolved the issue with the service provider yet, or do they still say that port 5060 is not open?  That issue needs to be fixed, but since you decline to provide any information about your network at that level we can't provide any advice on that aspect.

Earlier you said outbound calls worked, can you provide a debug for a working call?  Just "debug ccsip mess".

my debug ccsip messages:

 

*Apr 10 18:11:15.260: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:11468972595168471@76.80.72.220 SIP/2.0
Via: SIP/2.0/UDP 185.53.88.61:59245;branch=z9hG4bK1862106552
Max-Forwards: 70
From: <sip:209@76.80.72.220>;tag=1464036537
To: <sip:11468972595168471@76.80.72.220>
Call-ID: 1964042596-1860279529-1452998222
CSeq: 1 INVITE
Contact: <sip:209@185.53.88.61:59245>
Content-Type: application/sdp
Content-Length: 207
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH

2921LAB#User-Agent: Linksys-SPA942

v=0
o=209 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

*Apr 10 18:11:16.572: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:76.80.72.220 SIP/2.0
Via: SIP/2.0/UDP 69.59.30.22:5060;branch=z9hG4bK6a428fe5;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@69.59.30.22>;tag=as0ba523a8
To: <sip:76.80.72.220>
Contact: <sip:asterisk@69.59.30.22:5060>
Call-ID: 017d2e366e8fd51258597706333fd056@69.59.30.22:5060
CSeq: 102 OPTIONS
User-Agent: VYLmedia-SBCDAL
Date: Fri, 10 Apr 2020 18:02:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

*Apr 10 18:11:16.572: //66297/8141164B9159/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 69.59.30.22:5060;branch=z9hG4bK6a428fe5;rport
From: "asterisk" <sip:asterisk@69.59.30.22>;tag=as0ba523a8
To: <sip:76.80.72.220>;tag=2FF09C6C-9AF
Date: Fri, 10 Apr 2020 18:11:16 GMT
Call-ID: 017d2e366e8fd51258597706333fd056@69.59.30.22:5060
Server: Cisco-SIPGateway/IOS-15.7.3.M4
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp

2921LAB#uSupported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 371

v=0
o=CiscoSystemsSIP-GW-UserAgent 4379 8836 IN IP4 76.80.72.220
s=SIP Call
c=IN IP4 76.80.72.220
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 76.80.72.220
m=image 0 udptl t38
c=IN IP4 76.80.72.220
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

2921LAB#u
*Apr 10 18:11:18.092: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:9011441482455983@76.80.72.220 SIP/2.0
Via: SIP/2.0/UDP 185.53.88.36:54572;branch=z9hG4bK293856916
Max-Forwards: 70
From: <sip:5006@76.80.72.220>;tag=789326965
To: <sip:9011441482455983@76.80.72.220>
Call-ID: 240969301-1725008714-1011222116
CSeq: 1 INVITE
Contact: <sip:5006@185.53.88.36:54572>
Content-Type: application/sdp
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH

v=0
o=5006 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

*Apr 10 18:11:18.272: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:9011442037698349@76.80.72.220 SIP/2.0
Via: SIP/2.0/UDP 185.53.88.36:55090;branch=z9hG4bK1329605425
Max-Forwards: 70
From: <sip:2230@76.80.72.220>;tag=2043163711
To: <sip:9011442037698349@76.80.72.220>
Call-ID: 384477006-2105388998-1309220067
CSeq: 1 INVITE
Contact: <sip:2230@185.53.88.36:55090>
Content-Type: application/sdp
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH

v=0

2921LAB#u o=2230 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

*Apr 10 18:11:18.632: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:972595778361@76.80.72.220 SIP/2.0
To: 972595778361<sip:972595778361@76.80.72.220>
From: 209<sip:209@76.80.72.220>;tag=fcb0da24
Via: SIP/2.0/UDP 185.53.88.61:5070;branch=z9hG4bK-d82b9411b04ee763f8bc3a819ced3cf2;rport
Call-ID: d82b9411b04ee763f8bc3a819ced3cf2
CSeq: 1 INVITE
Contact: <sip:209@185.53.88.61:5070>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE
User-Agent: sipcli/v1.8
Content-Type: application/sdp
Content-Length: 279

v=0
o=sipcli-Session 18494007 1777193466 IN IP4 185.53.88.61
s=sipcli
c=IN IP4 185.53.88.61
t=0 0
m=audio 5072 RTP/AVP 18 0 8 101
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv

*Apr 10 18:11:18.820: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:11469972595168471@76.80.72.220 SIP/2.0
Via: SIP/2.0/UDP 185.53.88.61:61567;branch=z9hG4bK1534140086
Max-Forwards: 70
From: <sip:209@76.80.72.220>;tag=1743650323
To: <sip:11469972595168471@76.80.72.220>
Call-ID: 1398858069-31276121-1835386278
CSeq: 1 INVITE
Contact: <sip:209@185.53.88.61:61567>
Content-Type: application/sdp
Content-Length: 207
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH
User-Agent: Linksys-SPA942

v=0
o=209 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

*Apr 10 18:11:18.920: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:801146842002318@76.80.72.220 SIP/2.0
Via: SIP/2.0/UDP 45.151.255.178:53048;branch=z9hG4bK1202065564
Max-Forwards: 70
From: <sip:3600@76.80.72.220>;tag=1755593553
To: <sip:801146842002318@76.80.72.220>
Call-ID: 53082744-2055631046-1754975611
CSeq: 1 INVITE
Contact: <sip:3600@45.151.255.178:53048>
Content-Type: application/sdp
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH

v=0

2921LAB#u ao=3600 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

2921LAB#u all
*Apr 10 18:11:21.852: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:11470972595168471@76.80.72.220 SIP/2.0
Via: SIP/2.0/UDP 185.53.88.61:63425;branch=z9hG4bK1380828401
Max-Forwards: 70
From: <sip:209@76.80.72.220>;tag=1402649641
To: <sip:11470972595168471@76.80.72.220>
Call-ID: 1464816868-203176533-1771669524
CSeq: 1 INVITE
Contact: <sip:209@185.53.88.61:63425>
Content-Type: application/sdp
Content-Length: 207
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH
User-Agent: Linksys-SPA942


@moman62 wrote:

my debug ccsip messages:


Those do not show a working call, there are just a series of apparently unrelated Invites from various sources, no responses sent to any of them.

port 5060 is open and my ITSP acknowledges it.  I am attaching my dialpeers setup.

 

dial-peer voice 101 voip
description Incoming
session protocol sipv2
incoming called-number [2-9].........
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 102 voip
description Outbound Wan
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
session target ipv4:209.105.241.22
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 103 voip
description Inbound LAN
session protocol sipv2
incoming called-number 9T
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 104 voip
description Outbound LAN
destination-pattern 9.1[2-9].........
session protocol sipv2
session target ipv4:192.168.30.7
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte
no vad

Now you have the same interface bind on all the dial peers. Are you only using one interface on the SBC? If not you need to set the correct interface in the bind on the inbound and outbound dial peer for each direction. You should have one pair, in/outbound for the ITSP facing call leg, these should have the bind interface that is used for communication with the ITSP. Then you should have a pair for in/outbound to your phone system that should have the bind interface that is used to communicate with the internal phone system.



Response Signature


I am just trying to have inbound reach these extensions 33x and 417x internal calls look at my dialpeers setup and tell me where am I going wrong? I am using voip to voip connections only.

 

dial-peer voice 101 voip
description Incoming
session protocol sipv2
incoming called-number [2-9].........
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711ulaw
no vad


dial-peer voice 103 voip
description Inbound LAN
session protocol sipv2
incoming called-number 9T
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
no vad

 

dial-peer voice 102 voip

description Outbound Wan
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
session target ipv4:209.105.241.22
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte
no vad

 

dial-peer voice 104 voip
description Outbound LAN
destination-pattern 9.1[2-9].........
session protocol sipv2
session target ipv4:192.168.30.7
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte
no vad

 

I’m sorry its quite clear that you don’t have the needed skill set to get this to work and you constantly select to not give answer to questions you get. The information you do select to give is frankly often incomplete or inconsistent.

I recommend you to reach out to a certified parner to get onsite/IRL help with this as it’s quite likely that you will not be able to get to a working state with the help of this forum.



Response Signature