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Alcatel OminPCX SIP clients

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nbrown

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Hi, I'm trying to connect our Alcatel OminPCX phone system (192.168.100.235) to 3CX (192.168.100.194) as part of a trial. We don't have SIP trunks on the alcatel system but we do have SIP clients. I've created a new extension and added the connection details as a generic VoIP provider. Registration is OK and when I call this new extension (621) via an alcatel phone it rings a phone on 3CX perfectly OK. But when making a call from the 3CX extension ([email protected]) to an alcatel extension (611) nothing happens, just silence. I know the routes and rules are all OK and below is a section of the log file, can anyone help?

09-May-2014 10:32:53.155 Route to L:5.2[Line:10001>>611] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=7728fa3fcc3ef503 Call-ID=MThiYWM2N2U4NDdmZmRjYjk1NDgxMTM5M2U3YmU1NzM.:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-7728fa3fcc3ef503-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5061>
To: <sip:[email protected]:5060>
From: "621"<sip:[email protected]:5060>;tag=93121b39
Call-ID: MThiYWM2N2U4NDdmZmRjYjk1NDgxMTM5M2U3YmU1NzM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 413

v=0
o=3cxPS 255936430080 334386692097 IN IP4 192.168.100.194
s=3cxPS Audio call
c=IN IP4 192.168.100.78
t=0 0
m=audio 3000 RTP/AVP 0 8 18 9 18 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
09-May-2014 10:32:53.148 Outbound URI is used: sip:192.168.100.235:5060
09-May-2014 10:32:53.148 SLA is globally disabled
09-May-2014 10:32:53.146 Added leg L:5.2[Line:10001>>611]
09-May-2014 10:32:53.108 [Flow] Call(C:5): making call from L:5.1[Extn] to T:Line:10001>>611@[Dev:sip:[email protected]:5060]
09-May-2014 10:32:53.108 [CM503027]: Call(C:5): From: Extn:1000 ("Operator Extension" <sip:[email protected]:5061>) to T:Line:10001>>611@[Dev:sip:[email protected]:5060]
09-May-2014 10:32:53.108 [CM503004]: Call(C:5): Route 1: from L:5.1[Extn] to T:Line:10001>>611@[Dev:sip:[email protected]:5060]
09-May-2014 10:32:53.108 Line limit check: Current # of calls for line Lc:10001(@Alcatel[<sip:[email protected]:5060>]) is 0; limit is 2
09-May-2014 10:32:53.108 Call(C:5): Call from Extn:1000 to 611 matches outbound rule 'Rule for Sipgate (Nick) (internal 1)'
09-May-2014 10:32:53.108 [Flow] Call(C:5): has built target endpoint: Out#:>>Rule{Rule for Sipgate (Nick) (internal 1)}>>611 for call from L:5.1[Extn]
09-May-2014 10:32:53.108 [Flow] Target endpoint for 611 is Out#:>>Rule{Rule for Sipgate (Nick) (internal 1)}>>611
09-May-2014 10:32:53.108 Selected prefix: 6
09-May-2014 10:32:53.108 Looking for outbound rule: dialed = [611], processed: [611]; from-ext:
09-May-2014 10:32:53.108 [Flow] Building target endpoint to 611 from "Operator Extension" <sip:[email protected]:5061>
09-May-2014 10:32:53.108 [CM503010]: Call(C:5): Making route(s) from Extn:1000 to <sip:[email protected]:5061>
09-May-2014 10:32:53.108 Remote SDP is set for leg L:5.1[Extn]
09-May-2014 10:32:53.108 OnOffer from "Operator Extension"<sip:[email protected]:5061>;tag=8af1c41f90
09-May-2014 10:32:53.108 [CM505001]: Endpoint Extn:1000: Device info: Device Identified: [Man: Aastra;Mod: 5x series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, recvonly] UserAgent: [Aastra 55i/3.3.1.2235] PBX contact: [sip:[email protected]:5061]
09-May-2014 10:32:53.105 Inbound DID: ''; Phonebook Name: ''
09-May-2014 10:32:53.105 [CM500002]: Call(C:5): Info on incoming INVITE from Extn:1000:
Invite-IN Recv Req INVITE from 192.168.100.78:5060 tid=67727e5536af0a982.8d0cca12b4a68f69c Call-ID=be2881de189c5934:
INVITE sip:[email protected]:5061;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.78:5060;branch=z9hG4bK67727e5536af0a982.8d0cca12b4a68f69c
Max-Forwards: 70
Contact: "Operator Extension"<sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1A30B3>"
To: <sip:[email protected]:5061;user=phone>
From: "Operator Extension"<sip:[email protected]:5061>;tag=8af1c41f90
Call-ID: be2881de189c5934
CSeq: 14044 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="1000",realm="3CXPhoneSystem",nonce="414d535c097d31c443:20fff0e9af80367f997982d5161494cf",uri="sip:[email protected]:5061;user=phone",response="264b6e8bb1645fb4d80b73dbea052bab",algorithm=MD5
Supported: path, 100rel, replaces
User-Agent: Aastra 55i/3.3.1.2235
Allow-Events: aastra-xml, talk, hold, conference, LocalModeStatus
Content-Length: 382

v=0
o=MxSIP 0 1 IN IP4 192.168.100.78
s=SIP Call
c=IN IP4 192.168.100.78
t=0 0
m=audio 3000 RTP/AVP 0 8 18 9 18 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
09-May-2014 10:32:53.105 [CM503001]: Call(C:5): Incoming call from Extn:1000 to <sip:[email protected]:5061>
09-May-2014 10:32:53.096 Outbound URI is used: sip:[email protected]:5060;transport=udp
09-May-2014 10:32:53.096 IncomingCall: C:5 from <sip:[email protected]:5061> to <sip:[email protected]:5061>
09-May-2014 10:32:53.096 Added leg L:C:5.1[No endpoint yet]
09-May-2014 10:32:53.096 UasSession 407 started
09-May-2014 10:32:53.096 Call from "Operator Extension"<sip:[email protected]:5061>;tag=8af1c41f90 to <sip:[email protected]:5061;user=phone>;tag=84379a6f
 
What outbound rules do you have on the 3CX Phone System?
 
The call looks to be routing through OK but what happens later? The start and end of the log shows the same hour, minute, second, so where is the log after a "timeout" period when the call fails?

Have you used Wireshark to see what is sent, and if there is any response from the Alcatel PBX?
 
After about 20 seconds the call drops so you get a hang up tone, data below is the wireshark capture and below that is the whole log...

WIRESHARK
pPVfHEeddQOaINVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.194:5061;branch=z9hG4bK-d8754z-79423e745b451203-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5061>
To: <sip:[email protected]:5060>
From: "621"<sip:[email protected]:5060>;tag=e362662d
Call-ID: NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Proxy-Authorization: Digest username="621",realm="xnaim",nonce="d661cf04ac69320b25da6482828f2824",uri="sip:[email protected]:5060",response="85e653f7546025d6588a21126db88449",cnonce="647f3b311c04e32495685586decea460",nc=00000001,qop=auth,algorithm=MD5
Supported: replaces
User-Agent: 3CXPhoneSystem 12.0.35528.640 (34969)
Content-Length: 260

v=0
o=3cxPS 444210348032 218523238401 IN IP4 192.168.100.194
s=3cxPS Audio call
c=IN IP4 192.168.100.194
t=0 0
m=audio 7030 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

WHOLE LOG
12-May-2014 07:49:18.556 L:8.1[Extn]: Terminating targets, reason:
12-May-2014 07:49:18.556 Leg L:8.1[Extn] is terminated: Cause: BYE from PBX
12-May-2014 07:49:18.556 Terminated from "Operator Extension"<sip:[email protected]:5061>;tag=5df8454771 to <sip:[email protected]:5061;user=phone>;tag=8f466075; reason: Rejected
12-May-2014 07:49:18.556 L:8.1[Extn] Sending: OnSendResp Send 480/INVITE from 0.0.0.0:0 tid=859007c0f31f1f4af.68c3cc52f9fd10929 Call-ID=41a890b05251b9cb:
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.100.78:5060;branch=z9hG4bK859007c0f31f1f4af.68c3cc52f9fd10929
To: <sip:[email protected]:5061;user=phone>;tag=8f466075
From: "Operator Extension"<sip:[email protected]:5061>;tag=5df8454771
Call-ID: 41a890b05251b9cb
CSeq: 10288 INVITE
Warning: 499 CH3CX1.aimaviation.local "No answer"
Content-Length: 0
12-May-2014 07:49:18.556 SendMsg from <sip:[email protected]:5061;user=phone>;tag=8f466075 to "Operator Extension"<sip:[email protected]:5061>;tag=5df8454771
12-May-2014 07:49:18.518 ~Target=VoIPline:611@(Ln.10001@Alcatel)
12-May-2014 07:49:18.513 Call(C:8) is terminated
12-May-2014 07:49:18.507 [CM503020]: Call(C:8): Normal call termination. Call originator: Extn:1000. Reason: No answer
12-May-2014 07:49:18.507 [CM503016]: Call(C:8): Attempt to reach <sip:[email protected]:5061> from Extn:1000 has failed. Reason: No Answer
12-May-2014 07:49:18.507 [Flow] Current call diversion path:[]
12-May-2014 07:49:18.507 L:8.1[Extn] failed to reach Line:10001>>611, reason No Answer
12-May-2014 07:49:18.507 RerouteReq
12-May-2014 07:49:18.505 L:8.2[Line:10001>>611]: Terminating targets, reason: SIP ;cause=408 ;text="Request Timeout"
12-May-2014 07:49:18.505 Leg L:8.2[Line:10001>>611] is terminated: Cause: 408 Request Timeout/INVITE from local
12-May-2014 07:49:18.505 L:8.2[Line:10001>>611] got Terminated Recv 408/INVITE from 0.0.0.0:0 tid=79423e745b451203 Call-ID=NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.100.194:5061;branch=z9hG4bK-d8754z-79423e745b451203-1---d8754z-;rport
To: <sip:[email protected]:5060>;tag=7e096c49
From: "621"<sip:[email protected]:5060>;tag=e362662d
Call-ID: NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.
CSeq: 2 INVITE
Content-Length: 0
12-May-2014 07:49:18.505 Terminated from <sip:[email protected]:5060>;tag=7e096c49 to "621"<sip:[email protected]:5060>;tag=e362662d; reason: Error
12-May-2014 07:49:18.505 ~Route=Dev:sip:[email protected]:5060
12-May-2014 07:49:18.505 Call to T:Line:10001>>611@[Dev:sip:[email protected]:5060] from L:8.1[Extn] failed, cause: Cause: 408 Request Timeout/INVITE from local
12-May-2014 07:49:18.491 L:8.2[Line:10001>>611] got Failure: Failure Recv 408/INVITE from 0.0.0.0:0 tid=79423e745b451203 Call-ID=NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.100.194:5061;branch=z9hG4bK-d8754z-79423e745b451203-1---d8754z-;rport
To: <sip:[email protected]:5060>;tag=7e096c49
From: "621"<sip:[email protected]:5060>;tag=e362662d
Call-ID: NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.
CSeq: 2 INVITE
Content-Length: 0
12-May-2014 07:49:18.491 [CM503003]: Call(C:8): Call to <sip:[email protected]:5060> has failed; Cause: 408 Request Timeout/INVITE from local
12-May-2014 07:49:18.491 Session 69916 has failed in leg L:8.2[Line:10001>>611] ; Cause: 408 Request Timeout/INVITE from local
12-May-2014 07:49:18.491 Failure from <sip:[email protected]:5060>;tag=7e096c49 to "621"<sip:[email protected]:5060>;tag=e362662d
12-May-2014 07:48:57.820 Currently active calls - 1: [8]
12-May-2014 07:48:46.244 [CM503025]: Call(C:8): Calling T:Line:10001>>611@[Dev:sip:[email protected]:5060] for L:8.1[Extn]
12-May-2014 07:48:46.244 Route to L:8.2[Line:10001>>611] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=c747d776537b5e65 Call-ID=NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-c747d776537b5e65-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5061>
To: <sip:[email protected]:5060>
From: "621"<sip:[email protected]:5060>;tag=e362662d
Call-ID: NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 260

v=0
o=3cxPS 444210348032 218523238401 IN IP4 192.168.100.194
s=3cxPS Audio call
c=IN IP4 192.168.100.194
t=0 0
m=audio 7030 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
12-May-2014 07:48:46.234 Outbound URI is used: sip:192.168.100.235:5060
12-May-2014 07:48:46.234 SLA is globally disabled
12-May-2014 07:48:46.233 Added leg L:8.2[Line:10001>>611]
12-May-2014 07:48:46.195 [Flow] Call(C:8): making call from L:8.1[Extn] to T:Line:10001>>611@[Dev:sip:[email protected]:5060]
12-May-2014 07:48:46.195 [CM503027]: Call(C:8): From: Extn:1000 ("Operator Extension" <sip:[email protected]:5061>) to T:Line:10001>>611@[Dev:sip:[email protected]:5060]
12-May-2014 07:48:46.195 [CM503004]: Call(C:8): Route 1: from L:8.1[Extn] to T:Line:10001>>611@[Dev:sip:[email protected]:5060]
12-May-2014 07:48:46.195 Line limit check: Current # of calls for line Lc:10001(@Alcatel[<sip:[email protected]:5060>]) is 0; limit is 2
12-May-2014 07:48:46.195 Call(C:8): Call from Extn:1000 to 611 matches outbound rule 'Rule for Sipgate (Nick) (internal 1)'
12-May-2014 07:48:46.195 [Flow] Call(C:8): has built target endpoint: Out#:>>Rule{Rule for Sipgate (Nick) (internal 1)}>>611 for call from L:8.1[Extn]
12-May-2014 07:48:46.195 [Flow] Target endpoint for 611 is Out#:>>Rule{Rule for Sipgate (Nick) (internal 1)}>>611
12-May-2014 07:48:46.195 Selected prefix: 6
12-May-2014 07:48:46.195 Looking for outbound rule: dialed = [611], processed: [611]; from-ext:
12-May-2014 07:48:46.195 [Flow] Building target endpoint to 611 from "Operator Extension" <sip:[email protected]:5061>
12-May-2014 07:48:46.195 [CM503010]: Call(C:8): Making route(s) from Extn:1000 to <sip:[email protected]:5061>
12-May-2014 07:48:46.195 Remote SDP is set for leg L:8.1[Extn]
12-May-2014 07:48:46.195 OnOffer from "Operator Extension"<sip:[email protected]:5061>;tag=5df8454771
12-May-2014 07:48:46.195 [CM505001]: Endpoint Extn:1000: Device info: Device Identified: [Man: Aastra;Mod: 5x series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, recvonly] UserAgent: [Aastra 55i/3.3.1.2235] PBX contact: [sip:[email protected]:5061]
12-May-2014 07:48:46.193 Inbound DID: ''; Phonebook Name: ''
12-May-2014 07:48:46.193 [CM500002]: Call(C:8): Info on incoming INVITE from Extn:1000:
Invite-IN Recv Req INVITE from 192.168.100.78:5060 tid=859007c0f31f1f4af.68c3cc52f9fd10929 Call-ID=41a890b05251b9cb:
INVITE sip:[email protected]:5061;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.78:5060;branch=z9hG4bK859007c0f31f1f4af.68c3cc52f9fd10929
Max-Forwards: 70
Contact: "Operator Extension"<sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1A30B3>"
To: <sip:[email protected]:5061;user=phone>
From: "Operator Extension"<sip:[email protected]:5061>;tag=5df8454771
Call-ID: 41a890b05251b9cb
CSeq: 10288 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="1000",realm="3CXPhoneSystem",nonce="414d535c0980ffce09:944680bb9c98cea40fafce23b7e06411",uri="sip:[email protected]:5061;user=phone",response="0f1be8154ede47871416eeb781d3bd05",algorithm=MD5
Supported: path, 100rel, replaces
User-Agent: Aastra 55i/3.3.1.2235
Allow-Events: aastra-xml, talk, hold, conference, LocalModeStatus
Content-Length: 382

v=0
o=MxSIP 0 1 IN IP4 192.168.100.78
s=SIP Call
c=IN IP4 192.168.100.78
t=0 0
m=audio 3000 RTP/AVP 0 8 18 9 18 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
12-May-2014 07:48:46.193 [CM503001]: Call(C:8): Incoming call from Extn:1000 to <sip:[email protected]:5061>
12-May-2014 07:48:46.184 Outbound URI is used: sip:[email protected]:5060;transport=udp
12-May-2014 07:48:46.184 IncomingCall: C:8 from <sip:[email protected]:5061> to <sip:[email protected]:5061>
12-May-2014 07:48:46.184 Added leg L:C:8.1[No endpoint yet]
12-May-2014 07:48:46.183 UasSession 69907 started
12-May-2014 07:48:46.183 Call from "Operator Extension"<sip:[email protected]:5061>;tag=5df8454771 to <sip:[email protected]:5061;user=phone>;tag=8f466075
 
I've done a port mirror of the network interface for the phone system and performed a wireshark there too, all seems OK and the phone system is responding to the call as far as I can see, there's lots of INVITE's, a REGISTER and more INVITE's, not sure why it isn't working.
 
Does the alcatel have a firewall blocking any traffic/ports?
 
nbrown said:
12-May-2014 07:49:18.556 Terminated from "Operator Extension"<sip:[email protected]:5061>;tag=5df8454771 to <sip:[email protected]:5061;user=phone>;tag=8f466075; reason: Rejected[/quote]

Do you see the "rejected" reason in other call logs to the PBX? Are there logs in the Omni PBX to show what is going on with a call?
Click to expand...
 
The Alcatel has no firewalls between itself and the 3CX and always works fine with any SIP device such as X-lite or Aastra phone. Not sure if there are logs we can get to on the Alcatel PBX but I will find out.

Nick
 
There is no obvious logging I can find, unless someone can point me in the right direction!
 
I would use Wireshark and compare a call attempt from 3CX with a call from a set (perhaps an ATA if you have one handy) using the same extension number. Take note of any differences, even in the initial registration.

This just set-up as a Generic trunk, have you tried using a bridge trunk (slave)?
 
We did try comparing the Wireshark traces of a device that worked compared to one that didn't and although there were small differences there was nothing obvious that jumped out as being the problem. We'll try this again and maybe from a software client on the 3CX server as the comparison too. Our phone system provider have given us a quote for one of their engineers to visit and help with the diagnostics so we may have to take them up on this.

Yes this is setup as a generic VoIP provider not a trunk, the trunk just wouldn't work at all in either direction.
 
We might be getting somewhere, comparing a phone that works to the phone system that doesn't it appears as though 3CX may be trying to connect to the wrong port on the Alcatel system as in the invite request it has this...

User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: 65471 (65471)

Is there a way to force it to 5060?
 
OK, during registration with the Alcatel system this port 65471 appears (maybe NAT?) and I'm sure this is causing the issue. Is there any way to force registration to use port 5060, maybe using custom values in the outbound call? This also comes up in the SIP invites.

User Datagram Protocol, Src Port: sip (5060), Dst Port: 65471 (65471)
 
In the 3CX settings for the generic trunk you created, there should be a setting to tell 3CX the port to use at the far end (SIP Server Port). While most providers will use 5060 as a default, some don't so the ability to "customize" has to be there.
 
Try to set side trunk 3cx
Contact user part = originator caller id
 
We've got it working, in the end we have set up Gateways for outgoing calls (Had to un-tick 'Only Authenticated incoming calls' on the Alcatel system to get it to work too). We set the registration to 'Do not require' and we can now make outgoing calls only via these Gateways. We then set up Providers and these were used for incoming calls only.

It's a bit messy and you have to setup a lot of outbound rules but it's working now it allows us to test the 3CX system alongside our Alcatel Omnipcx which is what we wanted to achieve.

Thank you all for your help.
 
Glad you got it working. Unfortunately, integrating two PBX's, other than an interconnecting trunk for basic calls, is cumbersome at best, and a logistic nightmare at worst.

If the situation allows, and, of course, not all do, it's best to set-up a separate system using PSTN "test numbers", with VoIP trunking, or a gateway, that can then be integrated into in the final deployment.
 
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