Solved Direct SIP to Extension

[nbroad]

Hi,

I'm trying to enable direct SIP so that our 3rd party contact centre system (Genesys) can call 3CX extensions directly.
Prior to this I have tried a trunk - cannot call internal extensions over a trunk without DDI's. Why not?
I tried a bridge - Genesys cannot register with credentials so this won't work.

I have enabled direct SIP and added the extension number to the SIPID of my test extension (684).

I can successfully call 684 from a test SIP client (MicroSIP).
This is the invite from MicroSIP:

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.20.2.20:52080;rport;branch=z9hG4bKPj6fd541fb4cda437489e705408cb4acfc
Max-Forwards: 70
From: "Nigel MicroSIP" <sip:[email protected]>;tag=48887b1ec01e441eab1368fa3d96786f
To: <sip:[email protected]>
Contact: "Nigel MicroSIP" <sip:[email protected]:52080;ob>
Call-ID: 8430cf12e49f44dfb90fec13d56fe463
CSeq: 23869 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.19.17
Content-Type: application/sdp
Content-Length: 336

v=0
o=- 3772860939 3772860939 IN IP4 172.20.2.20
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 8 0 101
c=IN IP4 172.20.2.20
b=TIAS:64000
a=rtcp:4007 IN IP4 172.20.2.20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:154812105 cname:2b7156cf23e3268a


When I try and call 684 from the Genesys system I get 603 Decline from 3CX.
This is the invite from Genesys:

INVITE sip:[email protected]:5060 SIP/2.0
From: sip:[email protected]:5060;tag=FB4C5858-4861-4AB8-93A9-4D438BA804A4-292261
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 414
Content-Type: application/sdp
Via: SIP/2.0/UDP 172.28.2.169:5060;branch=z9hG4bKE7125CDF-77AE-479E-861F-5F45007737DC-27821
Contact: <sip:[email protected]:5060>
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
User-Agent: Genesys-SIPendpointSDK/9.0.004.07 (Windows 6.3.9600)
Max-Forwards: 70
X-Genesys-CallUUID: SLQSNP86ED20TBUPVBDH4VQFHC00000U
X-ISCC-CofId: location=eb_sip_switch;cofid=16777278
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: uui,100rel,timer


v=0
o=Genesys 1555036365 1 IN IP4 172.28.2.172
s=GSept-9.0.004.07
c=IN IP4 172.28.2.172
t=0 0
m=audio 9000 RTP/AVP 0 8 9 18 120 102 104 103 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=yes
a=rtpmap:120 opus/48000/2
a=rtpmap:102 iLBC/8000
a=rtpmap:104 iSAC/32000
a=rtpmap:103 iSAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

And the responses from 3CX:

03:21:55.797: SIPTR: Received [0,UDP] 332 bytes from 172.28.2.99:5060 <<<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.28.2.169:5060;branch=z9hG4bKE7125CDF-77AE-479E-861F-5F45007737DC-27821
To: <sip:[email protected]:5060>
From: sip:[email protected]:5060;tag=FB4C5858-4861-4AB8-93A9-4D438BA804A4-292261
Call-ID: [email protected]
CSeq: 1 INVITE
Content-Length: 0


03:21:55.845: SIPTR: Received [0,UDP] 407 bytes from 172.28.2.99:5060 <<<<<
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 172.28.2.169:5060;branch=z9hG4bKE7125CDF-77AE-479E-861F-5F45007737DC-27821
To: <sip:[email protected]:5060>;tag=574c820c
From: sip:[email protected]:5060;tag=FB4C5858-4861-4AB8-93A9-4D438BA804A4-292261
Call-ID: [email protected]
CSeq: 1 INVITE
Warning: 499 PNP3CX01.ezibuy.internal "Caller is forbidden"
Content-Length: 0

Can anyone spot the issue?

I note in the unsuccessful invite the To field is:
To: <sip:[email protected]:5060>
The IP address 172.28.2.169 is the Genesys SIP server - I'm not sure why it does this but apparently it's not something our vendors can change.


Compared to the successful invite from MicroSIP where the To field is:
To: <sip:[email protected]>

The unsuccessful invite has the first line:
INVITE sip:[email protected]:5060 SIP/2.0

The successful invite has the first line:
INVITE sip:[email protected] SIP/2.0

Any help would be appreciated.

Regards,
Nigel

 

[JohnS_3CX]

Hi Nigel,

In General Settings > Network > FQDN you have the option to enable this

Under each Extension > Options > Options > SIP ID there must be an ID assigned

I think you already reached this far, so now take a look at the guide here: https://www.3cx.com/docs/direct-sip/

But then the invite is addressed to itself (Genesys)

I note in the unsuccessful invite the To field is:
To: <sip:[email protected]:5060>
The IP address 172.28.2.169 is the Genesys SIP server - I'm not sure why it does this but apparently it's not something our vendors can change.

I think you answered your own question as per the above quote

 

[nbroad]

But then the invite is addressed to itself (Genesys)



I think you answered your own question as per the above quote

Thanks for your reply. You were correct.
Our Genesys vendor found the setting to set the host part in the TO field.
Once they set that to our FQDN, calls came in fine.
All resolved.

Thanks,
Nigel

 

[JohnS_3CX]

Hi Nigel, glad to hear it was resolved