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SIP Trunk between IP Office 500 V2 and 3CX

SIP Trunk between IP Office 500 V2 and 3CX

SIP Trunk between IP Office 500 V2 and 3CX

(OP)
Hi,

I have set up a SIP trunk on my Avaya that talks to a 3CX server on the same subnet *(using a 3CX bridge). I have three problems:

1. I get a successful trunk set up and can make a call from 3CX to an Avaya extension but audio from 3CX cannot be heard by the Avaya. When I packet capture I see below and then the rtp packets going to the public ip of the avaya (natted on a firewall) not the internal IP.

v=0
o=UserA 1133936427 1035663707 IN IP4 <AvayaPublicIP>
s=Session SDP
c=IN IP4 <AvayaPublicIP>
t=0 0
m=audio 49152 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

So my 3CX server is sending the RTP traffic to the public ip not internal because the Avaya is giving the incorrect IP address... Now the complication is that I assume it's getting the Public IP from the WAN IP address setting in the LAN settings of the Avaya BUT I also have an external SIP trunk for all of our calls so changing that setting will break inbound / outbound calls. Does anyone know any way of overriding the SDP settings per trunk in an IP Office?

2. I get no call coming to 3CX from the Avaya at all. I have set up a short code of 7N; and directed it to the required SIP trunk but my 3CX never sees the attempt when I dial 7<<3cx-ext>.
3. My SIP trunk to the outside world is on the Avaya so I would like to dial an outside line on 3cx and have it route through Avaya. I dial 7<externalnumber> but the call never completes.

Has anyone achieved this before?

Thanks

Graham

RE: SIP Trunk between IP Office 500 V2 and 3CX

1- set the "SIP Line >> Transport >> use network topology info to none.
2- Do you have Exts starts with 7 in avaya IPO? if not, run a monitor trace from IPO and see if it sends the invites.
3- I do not get it, you want to use an external line from other system and have it routed by the originator system? If you can provide and example of what you want to achieve it would be great!

RE: SIP Trunk between IP Office 500 V2 and 3CX

(OP)
Thanks icet500,

Yes of course, more detail!

So basically, we're phasing out the use of the Avaya IP Office 500 next year as it's end of contract and we can do things much cheaper and easier with a 3CX license (for which I have much more experience than an IP Office). For now, we're contracted and don't have the money (as a charity) to invest in both. We need to move quickly to get remote workers working smarter and again without buying licenses and additional phones to do so. I have built a 3cx pbx in the office and want to use that coupled with some SIP phones I have lying around or the soft phone to give to remote users. I don't want to be purchasing more SIP trunks for the 3CX software so it makes sense to trunk between 3cx and Avaya to allow each other to make internal calls and then route external calls from 3cx through the Avaya and to our existing provider.

Now, I have got further since posting this. I couldn't see how to amend the SDP settings so I have set up LAN2 on the internal lan also (this means I can stop using the public address field which was required o lan1) and set the SIP trunk up through that. That has cured most of the issues, I can now have a two way chat with someone from 3CX to Avaya and make outbound calls from 3cx through the Avaya, all by prepending the number with a "7" but I still cannot dial from the Avaya to 3CX with again no trace of anything hitting 3cx when I do a packet capture.

So to your questions, I have no extensions starting with 7 in the Avaya. I set up a shortcode as 7N; - Dial - N - 18 (My Line group for the 3cx trunk)
I wondered whether there is some sort of "Catch All" shortcode that was hitting the call first and sending it to my ARS where my providers SIP trunks are but I last saw an IP Office 20 years ago so things are a little different now! I haven't really used the Monitor app before so that's today's challenge. I will try the Network Topology, it's currently set to LAN2.

Thanks

Graham

RE: SIP Trunk between IP Office 500 V2 and 3CX

(OP)
OK, I see this:

58155410mS CMCallEvt: 0000000000000000 0.1298.0 -1 BaseEP: NEW CMEndpoint f1694680 TOTAL NOW=10 CALL_LIST=3
58155410mS CMCallEvt: 0000000000000000 0.1298.0 -1 Wes CSO.-1: NEW CMExtnEndpoint f1694680, Name=Wes CSO, Extn=231, Phys Extn=231
58155411mS CMCallEvt: CREATE CALL:67 (f16a2908)
58155411mS CMCallEvt: 0000000000000000 0.1299.0 -1 BaseEP: NEW CMEndpoint f1748e3c TOTAL NOW=11 CALL_LIST=3
58155413mS CMExtnEvt: Wes CSO: CMExtnHandler::SetCurrent( id: 0->1298 )
58155413mS CMCallEvt: 0a5b0a0400000512 0.1298.0 67 Wes CSO.0: StateChange: END=A CMCSIdle->CMCSDialInitiated
58155413mS CMExtnEvt: v=10 State, new=Dialling old=Idle,0,0,Wes CSO
58155414mS CMCallEvt: 0a5b0a0400000512 0.1298.0 67 Wes CSO.0: StateChange: END=A CMCSDialInitiated->CMCSDialling
58156885mS CMExtnEvt: v=(null) State, new=Idle old=Idle,0,0,231: Digit Key Pressed 7
58157138mS CMExtnEvt: v=(null) State, new=Idle old=Idle,0,0,231: Digit Key Pressed 4
58157297mS SIP Rx: UDP 10.91.10.91:50739 -> 10.91.10.4:5060

58157506mS CMExtnEvt: v=(null) State, new=Idle old=Idle,0,0,231: Digit Key Pressed 0
58157822mS CMExtnEvt: v=(null) State, new=Idle old=Idle,0,0,231: Digit Key Pressed 0
58158174mS CMExtnEvt: v=(null) State, new=Idle old=Idle,0,0,231: Digit Key Pressed 0
********** SysMonitor v11.0.0.2.0 build 23 [connected to 10.91.10.4 (Company)] **********
58196006mS PRN: Monitor Status IP 500 V2 11.0.0.2.0 build 23
58196006mS PRN: LAW=A PRI=0, BRI=4, ALOG=0, VCOMP=20, MDM=0, WAN=0, MODU=1 LANM=0 CkSRC=0 VMAIL=1(VER=2 TYP=3) 1-X=0 CALLS=2(TOT=67)
58197354mS SIP Rx: UDP 10.91.10.91:50739 -> 10.91.10.4:5060

58202667mS CMCallEvt: 0a5b0a0400000512 0.1298.0 67 Wes CSO.0: StateChange: END=A CMCSDialling->CMCSCompleted
58202667mS CMExtnEvt: v=10 State, new=PortRecoverDelay old=Dialling,0,0,Wes CSO
58202668mS CMLOGGING: CALL:2020/08/1411:01,00:00:00,000,231,O,74000,74000,WesCSO,,,1,,"",0,n/a
58202669mS CMExtnEvt: Wes CSO: CALL LOST (CMCauseNormal)
58202669mS CMExtnEvt: Wes CSO: Extn(231) Calling Party Number(231) Type(CMNTypeInternal)
58202669mS CMExtnEvt: Wes CSO: CMExtnHandler::SetCurrent( id: 1298->0 )
58202669mS CMCallEvt: 0a5b0a0400000512 0.1298.0 -1 Wes CSO.-1: StateChange: END=X CMCSCompleted->CMCSDelete
58202670mS CMCallEvt: 0000000000000000 0.1299.0 -1 BaseEP: DELETE CMEndpoint f1748e3c TOTAL NOW=3 CALL_LIST=1
58202670mS CMCallEvt: END CALL:67 (f16a2908)
58202670mS CMCallEvt: 0a5b0a0400000512 0.1298.0 -1 BaseEP: DELETE CMEndpoint f1694680 TOTAL NOW=2 CALL_LIST=1
58202672mS SIP Rx: UDP 10.91.10.91:5060 -> 10.91.10.4:5060

And that is it... I mean, I have a load of other chatter on but it all seems to be between other SIP devices, to be honest I have no idea how to separate the output on the monitor (I searched for all content containing the call id) and also not sure if maybe I need to tick something else to see more comprehensive capture in the preferences. We have 5 SIP 3rd party endpoint licenses but they work for a few days and then we get bugger all calls going in and out with an unauthorised and the only way I found to fix was delete and re-add the extension. That's why I started this whole endeavour. The company we pay for maintenance and support couldn't give a monkeys unless we pay lots of £££ or get their new hosted system (Gamma Horizon, spits on the ground in disgust). They stopped being so helpful when I managed to cut our monthly spend from £1200 to £400...

RE: SIP Trunk between IP Office 500 V2 and 3CX

Run monitor trace with SIP filter set to verbose and check send/receive options. Make a call to 3cx then stop the trace and search for the ext number you have used or by 3cx IP. Rx = receiving and Tx= sending out. So you should see and invite going out from that ext to your 3cx.

RE: SIP Trunk between IP Office 500 V2 and 3CX

Also, open system status and look for an alarm for that sip line. Or any other related.

RE: SIP Trunk between IP Office 500 V2 and 3CX

(OP)
Thanks, actually I now have a stage further, I have communication going but 3cX is rejecting it! Have posted on the 3cx community, I can't see anything wrong with it!

08/14/2020 6:01:14 PM - Currently active calls [none]
08/14/2020 6:01:12 PM - [CM102001]: Authentication failed for AuthFail Recv Req INVITE from 10.91.10.4:5060 tid=e9c3186d8e71a6b007275268a18d807d Call-ID=8cecd65b93e833be6963a7c595016ab5:
INVITE sip:4000@10.91.10.93 SIP/2.0
Via: SIP/2.0/UDP 10.91.10.4:5060;rport=5060;branch=z9hG4bKe9c3186d8e71a6b007275268a18d807d
Max-Forwards: 70
Contact: "10000" <sip:355@10.91.10.4:5060;transport=udp>
To: <sip:4000@10.91.10.93>
From: "10000" <sip:355@10.91.10.93>;tag=e6a8c5f2ca0315c2
Call-ID: 8cecd65b93e833be6963a7c595016ab5
CSeq: 1716009686 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, UPDATE
Content-Type: application/sdp
Proxy-Authorization: Digest username="10000",realm="3CXPhoneSystem",nonce="414d53595f36c2d495:0547f6591ffe29072d5f778621cd6c32",response="5de17ede80c40949c00881f44da66f02",uri="sip:4000@10.91.10.93"
Supported: timer
User-Agent: IP Office 11.0.0.2.0 build 23
P-Asserted-Identity: "10000" <sip:10000@10.91.10.4:5060>
P-Preferred-Identity: "10000" <sip:10000@10.91.10.4:5060>
Content-Length: 224

v=0
o=UserA 1295456529 1705827162 IN IP4 10.91.10.4
s=Session SDP
c=IN IP4 10.91.10.4
t=0 0
m=audio 49152 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
; Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
08/14/2020 6:01:12 PM - [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
Invite-UNK Recv Req INVITE from 10.91.10.4:5060 tid=e9c3186d8e71a6b007275268a18d807d Call-ID=8cecd65b93e833be6963a7c595016ab5:
INVITE sip:4000@10.91.10.93 SIP/2.0
Via: SIP/2.0/UDP 10.91.10.4:5060;rport=5060;branch=z9hG4bKe9c3186d8e71a6b007275268a18d807d
Max-Forwards: 70
Contact: "10000" <sip:355@10.91.10.4:5060;transport=udp>
To: <sip:4000@10.91.10.93>
From: "10000" <sip:355@10.91.10.93>;tag=e6a8c5f2ca0315c2
Call-ID: 8cecd65b93e833be6963a7c595016ab5
CSeq: 1716009686 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, UPDATE
Content-Type: application/sdp
Proxy-Authorization: Digest username="10000",realm="3CXPhoneSystem",nonce="414d53595f36c2d495:0547f6591ffe29072d5f778621cd6c32",response="5de17ede80c40949c00881f44da66f02",uri="sip:4000@10.91.10.93"
Supported: timer
User-Agent: IP Office 11.0.0.2.0 build 23
P-Asserted-Identity: "10000" <sip:10000@10.91.10.4:5060>
P-Preferred-Identity: "10000" <sip:10000@10.91.10.4:5060>
Content-Length: 224

v=0
o=UserA 1295456529 1705827162 IN IP4 10.91.10.4
s=Session SDP
c=IN IP4 10.91.10.4
t=0 0
m=audio 49152 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
08/14/2020 6:01:12 PM - [CM302002]: Authentication failed due to unidentified source of: SipReq: INVITE 4000@10.91.10.93 tid=e9c3186d8e71a6b007275268a18d807d cseq=1716009686 INVITE contact=355@10.91.10.4:5060 / 1716009686 from(wire)
08/14/2020 6:01:12 PM - Timezone offset: 60 min.

RE: SIP Trunk between IP Office 500 V2 and 3CX

(OP)
oh boy. I think i'm done for the day now as I just got two different results:

1. My Extension 355 which is a SIP client with a 3rd party endpoint license (which ironically is fully functioning today) rings 4000 and get's the results above.
2. a Digital extension tried to ring 4000 and it's silent, 3cx doesn't show any record of an attempt to communicate with it.

So I am not as far forward as I thought and for whatever reason the two calls are taking different approaches, I wonder if the call is going direct from the SIP client to 3CX after a broker from the Avaya (hence security error) whereas the digital handset has to use the phone system as the SIP client. I'll regroup next week when I am in the office and test again using the monitor as suggested. Thanks for all of your continued help and suggestions!

Thanks

Graham

RE: SIP Trunk between IP Office 500 V2 and 3CX

*Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
*Unidentified incoming call. Review INVITE and adjust source identification

These are the error messages you are getting in above trace. Also, What is ext 10000?

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