Solved Call from Multiple DID from provider DIDX goes to same ext

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tsww

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Hello Dear,

I am new to 3CX. I have multiple DID from DIDX and I am trying to test with 3CX. I did set up as per below.

- Inbound trunk setup
- Two ext 1000, and 1001
- two DID: DID1 and DID2
- I did set up Direct Inbound Dialing (DID) for DID1 to 1000 and for DID2 to 1001 under extension.
- I have created 2 inbound rules. DID1 route for 1000 and DID2 route to 1001.
- I have also configured at the provider side to DID1->1000@my3cx and DID2->1001

No matter what DID I call, call always go to ext 1001.

Did research and reviewed other posts with a similar issue and tried different settings under SIP Trunks>Inbound Parameters> Call Source Identification. But none is working. When I select "Use both "Call Source Identification" rules and "Caller Number/Name -> CalledNum" field mappings (Note: Disables catch all routing capability)"

I don't receive a call at all.

Does anybody use DIDX as a provider, would appreciate or know which settings would work.

Thanks,
Saur
 
You should check (confirm) the incoming calls in the 3CX Activity Log to see if the DID numbers sent, match your DID rules. If there is not an exact match, then the calls will route to the default destination. You may want to consider using wildcards (*) for the first few numbers in your rules, to cover variations of the numbers.
 
Last edited:
As per what @leejor mentioned, this is likely a matter of the PBX matching the incoming call, otherwise the trunk's "catch all" destination will be used since none of the Inbound Rules matched first.

These articles can help you better understand how the procedure works so you can manually tweak your settings. Note that as @cobaltit said, this will not be an issue with a 3CX supported provider since we test beforehand and prepare ready made templates that have been confirmed to work as intended.

https://www.3cx.com/docs/sip-trunk-inbound-calls/
https://www.3cx.com/docs/source-identification-issues/
 
See attached log, I have replaced the original IP for security reason. Call from DID come to 1001 ext but somehow it doesn't get processed to that ext and always goes to 1000 ext. also see attached settings of the inbound parameter. Please advise.
____________________________________________________________

Code:
12/22/2020 2:31:37 PM - Default best-route IP is determined as 142.93.146.233

12/22/2020 2:31:31 PM - L:18.1[Line:10001<<+14168009900] Sending: OnSendResp Send 180/INVITE from 0.0.0.0:0 tid=054f2e6d [email protected]:5060:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 159.203.27.198:5060;branch=z9hG4bK054f2e6d;rport=5060

Contact: <sip:[email protected]:5060>

To: <sip:[email protected]>;tag=6a2b9c31

From: <sip:[email protected]>;tag=as4437c06c

Call-ID: [email protected]:5060

CSeq: 102 INVITE

Content-Length: 0



12/22/2020 2:31:31 PM - SendMsg from <sip:[email protected]>;tag=6a2b9c31 to <sip:[email protected]>;tag=as4437c06c

12/22/2020 2:31:31 PM - L:18.2[Extn:1000] got Provisional Recv 180/INVITE from 127.0.0.1:5483 tid=5fca031a5ae2fa6e Call-ID=rcm9xsAPIVPVkqOmVSsXsQ..:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-524287-1---5fca031a5ae2fa6e;rport=5060

Contact: <sip:[email protected]:5483;rinstance=5e171fd4504dcbd8>

To: <sip:[email protected]>;tag=9e0aab0a

From: "+14168009900"<sip:[email protected]:5060;nf=e>;tag=f4b11439

Call-ID: rcm9xsAPIVPVkqOmVSsXsQ..

CSeq: 1 INVITE

User-Agent: 3CX Mobile Client

Content-Length: 0



12/22/2020 2:31:31 PM - Provisional(180) from <sip:[email protected]>;tag=9e0aab0a to "+14168009900"<sip:[email protected]:5060;nf=e>;tag=f4b11439

12/22/2020 2:31:31 PM - [CM505001]: Endpoint Extn:1000: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Mobile Client] PBX contact: [sip:[email protected]:5060]

12/22/2020 2:31:31 PM - [CM503002]: Call(C:18): Alerting Extn:1000 by contact <sip:[email protected]:5483/UDP>

12/22/2020 2:31:31 PM - Answer from <sip:[email protected]>;tag=9e0aab0a to "+14168009900"<sip:[email protected]:5060;nf=e>;tag=f4b11439

12/22/2020 2:31:31 PM - [CM503025]: Call(C:18): Calling T:Extn:1000@[Dev:sip:[email protected]:5483;rinstance=5e171fd4504dcbd8] for L:18.1[Line:10001<<+14168009900]

12/22/2020 2:31:31 PM - Route to L:18.2[Extn:1000] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=5fca031a5ae2fa6e Call-ID=rcm9xsAPIVPVkqOmVSsXsQ..:

INVITE sip:[email protected]:5483;rinstance=5e171fd4504dcbd8 SIP/2.0

Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---5fca031a5ae2fa6e;rport

Max-Forwards: 70

Contact: <sip:[email protected]:5060>

To: <sip:[email protected]>

From: "+14168009900"<sip:[email protected]:5060;nf=e>;tag=f4b11439

Call-ID: rcm9xsAPIVPVkqOmVSsXsQ..

CSeq: 1 INVITE

Alert-Info: <http://www.notused.invalidtld>;info=external

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE

Content-Type: application/sdp

Supported: replaces, timer

Content-Length: 421



v=0

o=3cxPS 28177985529970688 27226425039781889 IN IP4 127.0.0.1

s=3cxPS Audio call

c=IN IP4 127.0.0.1

t=0 0

m=audio 7056 RTP/AVP 0 8 9 3 18 112 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:3 GSM/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:112 opus/48000/2

a=fmtp:112 maxplaybackrate=16000;sprop-maxcapturerate=16000

a=rtpmap:101 telephone-event/8000

a=sendrecv

12/22/2020 2:31:31 PM - SLA is globally disabled

12/22/2020 2:31:31 PM - Added leg L:18.2[Extn:1000]

12/22/2020 2:31:31 PM - [Flow] Call(C:18): making call from L:18.1[Line:10001<<+14168009900] to T:Extn:1000@[Dev:sip:[email protected]:5483;rinstance=5e171fd4504dcbd8]

12/22/2020 2:31:31 PM - [CM503027]: Call(C:18): From: Line:10001<<+14168009900 ("+14168009900" <sip:[email protected]:5060>)  to  T:Extn:1000@[Dev:sip:[email protected]:5483;rinstance=5e171fd4504dcbd8]

12/22/2020 2:31:31 PM - [CM503004]: Call(C:18): Route 1: from L:18.1[Line:10001<<+14168009900] to T:Extn:1000@[Dev:sip:[email protected]:5483;rinstance=5e171fd4504dcbd8]

12/22/2020 2:31:31 PM - [Flow] Endpoint Extn:1000 has no forwarding rule on reason 'All calls'

12/22/2020 2:31:31 PM - [Flow] No office hours set, office hours assumed

12/22/2020 2:31:31 PM - [Flow] Call(C:18): has built target endpoint: Extn:1000 for call from L:18.1[Line:10001<<+14168009900]

12/22/2020 2:31:31 PM - [Flow] Target endpoint for 1000 is Extn:1000

12/22/2020 2:31:31 PM - [Flow] Building target endpoint to 1000 from "+14168009900" <sip:[email protected]:5060>

12/22/2020 2:31:31 PM - [CM503010]: Call(C:18): Making route(s) from Line:10001<<+14168009900 to <sip:[email protected]:5060/UDP>

12/22/2020 2:31:31 PM - Remote SDP is set for leg L:18.1[Line:10001<<+14168009900]

12/22/2020 2:31:31 PM - [CM505003]: Provider:[ca.didx.net] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]

12/22/2020 2:31:31 PM - Inbound DID: ''; Phonebook Name: ''

12/22/2020 2:31:31 PM - [CM500002]: Call(C:18): Info on incoming INVITE from Line:10001<<+14168009900:

Invite-IN Recv Req INVITE from 159.203.27.198:5060 tid=054f2e6d [email protected]:5060:

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 159.203.27.198:5060;branch=z9hG4bK054f2e6d;rport=5060

Max-Forwards: 70

Contact: <sip:[email protected]:5060>

To: <sip:[email protected]>

From: <sip:[email protected]>;tag=as4437c06c

Call-ID: [email protected]:5060

CSeq: 102 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Content-Type: application/sdp

Date: Tue, 22 Dec 2020 19:22:34 GMT

Supported: replaces, timer

User-Agent: DiDXsuPErTecSIPCa

Remote-Party-ID: "+14168009900" <sip:[email protected]>;party=calling;privacy=off;screen=no

Content-Length: 305



v=0

o=root 180771332 180771332 IN IP4 199.47.126.6

s=Asterisk PBX 11.21.0

c=IN IP4 199.47.126.6

t=0 0

m=audio 49078 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

12/22/2020 2:31:31 PM - [CM503001]: Call(C:18): Incoming call from Line:10001<<+14168009900 to <sip:[email protected]:5060>

12/22/2020 2:31:31 PM - Line limit check: Current # of calls for line Lc:10001(@ca.didx.net[<sip:[email protected]:5060/UDP>]) is 1; limit is 10

12/22/2020 2:31:31 PM - Blacklist check: number '+14168009900', list: ''; result = false

12/22/2020 2:31:31 PM - Dev(312647404):[sip:[email protected]:5060 / 10001]: PBX contact is public IP: <sip:[email protected]:5060/UDP>

12/22/2020 2:31:31 PM - [CM503012]: Inbound office hours rule (unnamed) for 10001 forwards to DN:1000

12/22/2020 2:31:31 PM - [Flow] No office hours set, office hours assumed

12/22/2020 2:31:31 PM - [Flow] Looking for inbound target: called=1001; caller="+14168009900" <sip:+14168009900@:0>

12/22/2020 2:31:31 PM - CallerNameAddr: "+14168009900"<sip:+14168009900;nf=e>

12/22/2020 2:31:31 PM - No inbound caller ID reformat rule for DN:10001 is defined, or it is disabled (<Rules />)

12/22/2020 2:31:31 PM - Source is identified as trunk Lc:10001(@ca.didx.net[<sip:[email protected]:5060/UDP>])

12/22/2020 2:31:31 PM - IncomingCall: C:18 from <sip:[email protected]:0/UDP> to <sip:[email protected]:0/UDP>

12/22/2020 2:31:31 PM - Added leg L:C:18.1[No endpoint yet]

12/22/2020 2:31:31 PM - Call from <sip:[email protected]>;tag=as4437c06c to <sip:[email protected]>;tag=6a2b9c31

12/22/2020 2:31:31 PM - Source is identified as trunk Lc:10001(@ca.didx.net[<sip:[email protected]:5060/UDP>])

12/22/2020 2:31:26 PM - Default best-route IP is determined as 142.93.146.233

12/22/2020 2:31:24 PM - Lc:10000(@VoIP.MS[<sip:mad::0/UDP>]) Resolving targets: primary - sip:[Check 3CX VoIP.ms Configuration Guide]:5060, no secondary over +UDP+IPv4+IPv6
 

Attachments

  • call source identification.JPG
    call source identification.JPG
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Does called number is needed because I have set forwarding from DIDX side to
DID1 to [email protected] and DID2 [email protected], and it actually sending it as per log. Why 3cx don't forward call to ext as per its coming.
 
Hi tsww,

Try with changing the following:

1.
SIP Trunks => DIDX (Name of trunk) => General
Main Trunk No: DIDX (Name of trunk)
Destination for calls during office hours: End Call
Destination for calls outside office hours: End Call

2.
SIP Trunks => DIDX (Name of trunk) => DID's
Add DID1 and DID1

3.
Inbound Rules => DID1 Route calls to 1000
Inbound Rules => DID2 Route calls to 1001
 
Hi tsww,

Try with changing the following:

1.
SIP Trunks => DIDX (Name of trunk) => General
Main Trunk No: DIDX (Name of trunk)
Destination for calls during office hours: End Call
Destination for calls outside office hours: End Call

2.
SIP Trunks => DIDX (Name of trunk) => DID's
Add DID1 and DID1

3.
Inbound Rules => DID1 Route calls to 1000
Inbound Rules => DID2 Route calls to 1001
Hi Ivan, I have done exactly what you have suggested see the below result in the log. It doesn't work.
___________________________________________________________________________________
12/26/2020 12:33:42 PM - Default best-route IP is determined as 142.93.146.233
12/26/2020 12:33:41 PM - Currently active calls [none]
12/26/2020 12:33:40 PM - Leg L:29.1[Line:10000<<+14168009900] is terminated: Cause: BYE from PBX
12/26/2020 12:33:40 PM - Terminated from <sip:[email protected]>;tag=as4526b2ac to <sip:[email protected]>;tag=b7ff3466; reason: Rejected
12/26/2020 12:33:40 PM - L:29.1[Line:10000<<+14168009900] Sending: OnSendResp Send 486/INVITE from 0.0.0.0:0 tid=68a7ef36 [email protected]:5060:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 159.203.27.198:5060;branch=z9hG4bK68a7ef36;rport=5060
To: <sip:[email protected]>;tag=b7ff3466
From: <sip:[email protected]>;tag=as4526b2ac
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Warning: 499 K803G-STUDI-UOIYI-GGHHS-1LCMF.localdomain "Disabled by Forwarding Rule"
Content-Length: 0

12/26/2020 12:33:40 PM - SendMsg from <sip:[email protected]>;tag=b7ff3466 to <sip:[email protected]>;tag=as4526b2ac
12/26/2020 12:33:40 PM - Call(C:29) is terminated
12/26/2020 12:33:40 PM - [CM503020]: Call(C:29): Normal call termination. Call originator: Line:10000<<+14168009900. Reason: Disabled by Forwarding Rule
12/26/2020 12:33:40 PM - [CM503016]: Call(C:29): Attempt to reach <sip:@142.93.146.233:5060> from Line:10000<<+14168009900 has failed. Reason: Busy
12/26/2020 12:33:40 PM - [Flow] Call(C:29): has built target endpoint: EndCall:EndCall for call from L:29.1[Line:10000<<+14168009900]
12/26/2020 12:33:40 PM - [Flow] Call(C:29): building EndCall endpoint (routing rule)
12/26/2020 12:33:40 PM - [CM503010]: Call(C:29): Making route(s) from Line:10000<<+14168009900 to <sip:@142.93.146.233:5060/UDP>
12/26/2020 12:33:40 PM - Remote SDP is set for leg L:29.1[Line:10000<<+14168009900]
12/26/2020 12:33:40 PM - [CM505003]: Provider:[DIDX] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
12/26/2020 12:33:40 PM - Inbound DID: ''; Phonebook Name: ''
12/26/2020 12:33:40 PM - [CM500002]: Call(C:29): Info on incoming INVITE from Line:10000<<+14168009900:
Invite-IN Recv Req INVITE from 159.203.27.198:5060 tid=68a7ef36 [email protected]:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 159.203.27.198:5060;branch=z9hG4bK68a7ef36;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=as4526b2ac
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Content-Type: application/sdp
Date: Sat, 26 Dec 2020 17:24:38 GMT
Supported: replaces, timer
User-Agent: DiDXsuPErTecSIPCa
Remote-Party-ID: "+14168009900" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Length: 305

v=0
o=root 387493850 387493850 IN IP4 199.47.126.6
s=Asterisk PBX 11.21.0
c=IN IP4 199.47.126.6
t=0 0
m=audio 33940 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
12/26/2020 12:33:40 PM - [CM503001]: Call(C:29): Incoming call from Line:10000<<+14168009900 to <sip:@142.93.146.233:5060>
12/26/2020 12:33:40 PM - Line limit check: Current # of calls for line Lc:10000(@DIDX[<sip:[email protected]:5060/UDP>]) is 1; limit is 10
12/26/2020 12:33:40 PM - Blacklist check: number '+14168009900', list: ''; result = false
12/26/2020 12:33:40 PM - Dev(1516214074):[sip:[email protected]:5060 / 10000]: PBX contact is public IP: <sip:[email protected]:5060/UDP>
12/26/2020 12:33:40 PM - [CM503012]: Inbound office hours rule (unnamed) for 10000 forwards to DN:
12/26/2020 12:33:40 PM - [Flow] No office hours set, office hours assumed
12/26/2020 12:33:40 PM - [Flow] Looking for inbound target: called=1001; caller="+14168009900" <sip:+14168009900@:0>
12/26/2020 12:33:40 PM - CallerNameAddr: "+14168009900"<sip:+14168009900;nf=e>
12/26/2020 12:33:40 PM - No inbound caller ID reformat rule for DN:10000 is defined, or it is disabled (<Rules />)
12/26/2020 12:33:40 PM - Source is identified as trunk Lc:10000(@DIDX[<sip:[email protected]:5060/UDP>])
12/26/2020 12:33:40 PM - IncomingCall: C:29 from <sip:[email protected]:0/UDP> to <sip:[email protected]:0/UDP>
12/26/2020 12:33:40 PM - Added leg L:C:29.1[No endpoint yet]
12/26/2020 12:33:40 PM - Call from <sip:[email protected]>;tag=as4526b2ac to <sip:[email protected]>;tag=b7ff3466
12/26/2020 12:33:40 PM - Source is identified as trunk Lc:10000(@DIDX[<sip:[email protected]:5060/UDP>])
12/26/2020 12:33:40 PM - Lc:10002(@VoIP.MS[<sip:@:0/UDP>]) Resolving targets: primary - sip:[Check 3CX VoIP.ms Configuration Guide]:5060, no secondary over +UDP+IPv4+IPv6
 
tsww,

Based on what is the configuration DIDX that provided could be helpful.

I see on DIDX website that for Asterisk setup they don't require any kind of authentication, you just need to match default context in sip.conf where the inbound calls will land.
Contexts as such don't exist in 3cx.

Try with creating two different trunks against DIDX with each DID you have with them, and then route to 1001 and 1002.
 
tsww,

Based on what is the configuration DIDX that provided could be helpful.

I see on DIDX website that for Asterisk setup they don't require any kind of authentication, you just need to match default context in sip.conf where the inbound calls will land.
Contexts as such don't exist in 3cx.

Try with creating two different trunks against DIDX with each DID you have with them, and then route to 1001 and 1002.
I did try as you suggested, but it's going to ext set in the first trunk. In other words, DID1 and DID2 both land to Ext set in the first trunk.
 
Finally, It worked now with 2 different trunks and an inbound parameter set with the custom setting as per the attached screenshot. Thanks, everyone, for your help.

I created a separate trunk for each DID, and In the custom field of the inbound parameter for each tunk, I put an extension number for that DID. Also, on DIDX (DID provider) side, I have set a call to land on that same extension that I have set for a particular DID.

I hope this will help someone and will save time for him.
 

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