Integration Between 3CX and Elastix?

Hello Masters,

Thanks in advance,

I have 3CX system at my Corporate office, and in branch office i have Asterisk.
every thing is working fine on both side individually.

We have lease line between corporate office to Branch office.

now, my management gave me instruction to integrate these both so that we able to call from corporate office to branch office locally. i tried to do the setup but could not succeed.

NOTE:Corporate office extension 100,101,102 and so on.
3cx ip address is 192.168.14.5
2. branch office extension 600,601,602 and on on.
asterisk ip address is 192.168.101.2

if some one dial 605 should press 9 first and the extension no like 9605 should reach on the same extension and vice versa.

if you could able to send me any link where step by step configuration mentioned. i would be really thankfull to you guys. kindly help me to come out from this situation.

I would suggest looking at either a 3CX or a Elastix forum for assistance.

It is unlikely users here would know the details of either product as well as frequenters of those sites.

thanks for the reply john,

i able to do now, the thing is that from corporate office i am able to call
but from branch office i am not able to call. when i try to call from branch office, on 3cx logs shows:-

[CM302002]: Authentication failed due to unidentified source of: SipReq:  INVITE 151@192.168.14.5 tid=17e06a77 cseq=INVITE contact=600@192.168.101.2:5060 / 103 from(wire)
06-Dec-2016 12:22:38.652	[CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
			Invite-UNK Recv Req INVITE from 192.168.101.2:5060 tid=17e86b6c Call-ID=5022cfe50058ef760625f70f1779cb66@192.168.101.2:5060:
			INVITE sip:151@192.168.14.5 SIP/2.0
			Via: SIP/2.0/UDP 192.168.101.2:5060;branch=z9hG4bK17e86b6c
			Max-Forwards: 70
			Contact: <sip:600@192.168.101.2:5060>
			To: <sip:151@192.168.14.5>
			From: "Reception"<sip:600@192.168.101.2>;tag=as7c491650
			Call-ID: 5022cfe50058ef760625f70f1779cb66@192.168.101.2:5060
			CSeq: 102 INVITE
			Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
			Content-Type: application/sdp
			Date: Tue, 06 Dec 2016 12:22:38 GMT
			Supported: replaces, timer
			User-Agent: FPBX-2.11.0(11.24.0)
			Content-Length: 285
			
			v=0
			o=root 1894246311 1894246311 IN IP4 192.168.101.2
			s=Asterisk PBX 11.24.0
			c=IN IP4 192.168.101.2
			t=0 0
			m=audio 19250 RTP/AVP 8 3 0 101
			a=rtpmap:8 PCMA/8000
			a=rtpmap:3 GSM/8000
			a=rtpmap:0 PCMU/8000
			a=rtpmap:101 telephone-event/8000
			a=fmtp:101 0-16
			a=ptime:20
			a=sendrecv

can you pls tell what should i do?

That appears to be a message/error from 3CX.

Since that is a closed source, proprietary system, you’ll need to contact that company for support.

As this is a forum for Asterisk and not for 3CX you may not find many people here who have done integration like you are doing. I’d suggest trying other forums where people may have more experience.

thanks Jcolp and Mjordan. i do that thaks for reply.

i sugest test following

  1. Branch office 3CX softphone able registering with 3CX
  2. From menu trunk PSTN 3CX try registering with SIP Ext. from elastix

Elastix is dead.

Regards

1 Like

ELASTIX SOLD OUT & BIT THE DUST… It’s Now 3CX

For the Continuation of us Elastix users It is now the “Isabel” project…

Thanks

RIP Elastix. :skull_crossbones: