3CX-Cisco CME SIP Trunk

[aws.dabbagh]

hi all,

I've been trying to create a SIP trunk between the 3CX and the CME we have in our network so callers on each system can reach each other.

the config I'm using starts with:

sip-ua
authentication username 10006 password 7 050D175C75424A0C

that authenticates the tunnel ID (from 3CX) and password (also from 3CX bridge creation page) with the Cisco router.

I immediately receive an error that the CME router was blacklisted because "PBX: blocked for too many failed authentications; User-Agent: Cisco-SIPGateway/IOS"

I'm using the username (ID) and password copied from the bridge trunk page in 3CX server itself.

what am I doing wrong? do you have any suggestions?

thanks all

 

[AWS2P]

Hi, not sure 3CX bridge is able to connect 3CX to Cisco , Bridge feature is done to inter-connect 3CX PBXs, so give a try with generic sip trunk

 

[cobaltit]

Suggestions, yes. Whitelist the IP for one, and then fire up PCAP and look at the 3CX activity logs

 

[aws.dabbagh]

thanks all,

I'm using the

Hi, not sure 3CX bridge is able to connect 3CX to Cisco , Bridge feature is done to inter-connect 3CX PBXs, so give a try with generic sip trunk

thanks for your reply, aws2p. my scenario is:

3CX server is connected to outside using SIP trunks.

3CX--->international PBX

I want to implement a bridge trunk to allow CME users to make national/international calls.
CME--->3CX--->international PBX

I can't make a direct connection.

will a generic SIP trunk allow that? any information about how I can configure it?

thanks again for support.

 

[aws.dabbagh]

Suggestions, yes. Whitelist the IP for one, and then fire up PCAP and look at the 3CX activity logs

thanks for the reply, cobaltit

I followed your suggestions, found that the CME keeps sending REGISTER requests without any reply from the 3CX, and the Bridge Trunk is not coming up.

any idea what that is happening? output follows:

Session Initiation Protocol (REGISTER)
Request-Line: REGISTER sip:10.10.30.26:5090 SIP/2.0
Method: REGISTER
Request-URI: sip:10.10.30.26:5090
Request-URI Host Part: 10.10.30.26
Request-URI Host Port: 5090
[Resent Packet: True]
[Suspected resend of frame: 26574]
Message Header
Via: SIP/2.0/UDP 10.10.30.6:5060;branch=z9hG4bK130711261
From: <sip:[email protected]>;tag=17F1DFA4-1C45
SIP from address: sip:[email protected]
SIP from address User Part: 10004
SIP from address Host Part: 10.10.30.26
SIP from tag: 17F1DFA4-1C45
To: <sip:[email protected]>
SIP to address: sip:[email protected]
Date: Tue, 19 Nov 2019 10:28:12 GMT
Call-ID: 9F42B73D-9ED11EA-802793E7-E0F7021D
[Generated Call-ID: 9F42B73D-9ED11EA-802793E7-E0F7021D]
User-Agent: Cisco-SIPGateway/IOS-15.5.1.T2
Max-Forwards: 70
Timestamp: 1574159292
CSeq: 3 REGISTER
Sequence Number: 3
Method: REGISTER
Contact: <sip:[email protected]:5060>
Contact URI: sip:[email protected]:5060
Contact URI User Part: 10004
Contact URI Host Part: 10.10.30.6
Contact URI Host Port: 5060
Expires: 3600
Supported: path
Content-Length: 0

 

[AWS2P]

Sorry , i'm not experienced on your specific scenario, wait for others forumer or 3CX team with better advice

 

[aws.dabbagh]

thanks anyway, aws2p. your support is appreciated.

 

[JohnS_3CX]

Hi @aws.dabbagh

This is not something that is directly supported and that I can help you with, but you may still be able to get it to work if you spend some time on it.

Take a look at this Asterisk guide, the part that should interest you the most is at the bottom, which shows you what invite the system expects to receive. If you are able to make it match you may be able to get it to work.

https://www.3cx.com/docs/

 

[aws.dabbagh]

hi all,

after a few days of trying, I got some progress. the issue that remains is authentication.

as a trial:

1) I set up a SIP trunk on 3CX, without authentication, then directed all calls to my extension.

2) I set up an outbound rule with an call with a prefix of "3" directed out the SIP trunk.

3) on the CME, I configured the below:

!
dial-peer voice 101 voip
description for incoming calls from 3CX
session protocol sipv2
session target ipv4:10.10.30.26
!
dial-peer voice 102 voip
description for outgoing calls to 3CX
destination-pattern 3...
session protocol sipv2
session target ipv4:10.10.30.26
codec g711ulaw
!

4) after all that, I can call any extension 3CX --> CME. but from CME --> 3CX any number I call is redirected to my extension because of the config in point (1).

when I enable authentication, the connection is closed and the CME IP is blacklisted for too many failed authentications. ( I know I can follow "cobaltit" suggestion and whitelist it. but I want to reach a point when it is authenticated first.)

I've tried the following config for authentication, but it's not working:

!
dial-peer voice 1 voip
description this is necessary to register with 3CX
destination-pattern 10004
!
!
sip-ua
credentials username 10004 password 7 00554155500E5D realm 3CXPhoneSystem
authentication username 10004 password 7 06575D72181B5F
no remote-party-id
registrar ipv4:10.10.30.26 expires 3600
sip-server ipv4:10.10.30.26
!
any guidance?

 

[leejor]

Going back through the forums, there have been quite a few people tying to achieve this (Cisco to 3CX interconnection) using Bridge, or Generic SIP trunks. Unfortunately, I can;'t find one post (there may be, but I didn't come across it), of it working successfully. In another forum, there was a suggestion to use an IP to IP gateway, but nothing about whether that was tried, and worked.

 

[aws.dabbagh]

hi all,

I think there is a way to work around that leejor.

current situation is as follows:

3CX--> CME is working
CME-->3CX getting busy tone

no logs on "debug ccsip message" nor on "debug ccsip non-call"

3CX keeps sending SIP error "407 Proxy authentication required" for any call INVITE sent. that is based on Packet Capture from 3CX server. which means the calls are reaching the 3CX server.

how do I get the CME to send authentication?

thanks