Avaya - 3CX Bridge Scenario using Avaya V11

[gbrown100]

Hiya,

I have followed several documents on linking via SIP trunks which have failed, I then followed this: https://www.nazaudy.com/index.php/9...hone-system-vs-avaya-ip-500#BOTTOMSetupBridge which got me to have a Bridge between Avaya IP Office 500 V2 and 3CX but the screen shots are from an old release and I'm not 100% sure I am getting it right!

I have a public cloud hosted 3CX and a local (behind NAT) Avaya, ports are unrestricted between destinations but from 3CX to Avaya I get an established call but no audio and from the Avaya to 3CX it just disappears and does nothing. It's the SIP URI settings that have changed dramatically on the Avaya V11 software.

Does anyone have any notes or info on the V11 software release and getting a trunk set up between them?

Thanks

Graham

 

[leejor]

Did you happen to come across this site?

https://www.claudion.com/3cx-trunk-avaya-ip-office-500-v2/

 

[gbrown100]

Thanks leejor, that's the one I tried first! No joy at all on that one. Both examples use local instances as opposed to a cloud 3cx instance so I wonder if there's a NAT problem. I'm going to build a local instance and see what happens.

 

[leejor]

As long as all of the correct ports are open, I can't see that there should be a difference, as in either case, they are are "remote" to the Avaya.

 

[gbrown100]

Well, I have got further using an internal server...

I have made a call from a 3cx App through to an extension in the Avaya, I could hear the audio from the Avaya but they could not hear me, a call to an outside line from 3cx via the Avaya resulted in no audio either way. Now, prior to this it wouldn't even ring but when checking the 3CX logs I found it was trying the ext@<avayaexternalIP> not ext@<avayainternalIP> despite the trunk specifying the internal IP to the Avaya. My IP Office has an outbound trunk to Gamma so I guess a setting somewhere defaults to that. I overrode this in the ITSP and Local Domain in the SIP Tab of the trunk on the Avaya to get to this point.

I have noticed in the log I still see this:

08/12/2020 3:14:16 PM - L:8.2[Line:10000>>221] got Provisional Recv 180/INVITE from <avayinternalip>:5060 tid=24b25607f8ba6165 Call-ID=dbn6LC3q9fprlKviPLZZuA..: SIP/2.0 180 Ringing Via: SIP/2.0/UDP <3cxinternalip>:5060;branch=z9hG4bK-524287-1---24b25607f8ba6165;rport Contact: <sip:221@<avayainternalip>:5060;transport=udp> To: <sip:221@<avayainternalip>:5060;transport=udp>;tag=20c8303fbdc1f027 From: "Graham Brown" <sip:10000@<avayainternalip>:5060;transport=udp>;tag=dc191a54 Call-ID: dbn6LC3q9fprlKviPLZZuA.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, UPDATE Server: IP Office 11.0.0.2.0 build 23 Supported: timer P-Asserted-Identity: <sip:*@<AvayaEXTERNALip>:5060> P-Preferred-Identity: Unknown <sip:*@<avayaEXTERNALip>:5060> Content-Length: 0

So we're still getting the external IP in the traffic and it's to do with the SIP URI's in the Avaya... I'm remote from the office today and can't keep risking irritating people by making test calls but any thoughts from anyone as to how to format the URI's for the avaya would be most welcome!

PS - I should add, I know it's not supported, this is for a charity suckered into a 7 year contract for this system that ends in May next year. They need remote working now and don't have the budget to be replacing the system whilst they're still paying for it and no point buying hardware they are going to replace next year...

 

[gbrown100]

Update for today, a packet capture form 3cx shows it sending the RTP packets to the (natted) public ip of the Avaya not the local IP set up in the trunk. This is why, not idea how to modify the SDP section from the Avaya though and that's not a 3CX issue!

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

 

[leejor]

If you are using a Bridge trunk at the 3CX end, then I'm going to assume it is set as a master, to allow the Avaya to register. When the Avaya registers, it will tell 3Cx the IP it wishes to be "contacted" on.(check a registration to confirm this) As (I assume), the Avaya has a public IP, then that is the one that 3CX will be "communicating" with. As with any remote devices (STUN, SBC) connected to 3CX, the PBX should not be able to contact local IPs, behind a router, directly. It would be using the Public IP plus the port number. Using port forwarding, the remote router uses the port number to identify which local IP, is to be the destination. If port 5060 is currently in use at the Avaya end, then it should be registering with a different port.

I'm not certain if the problems are going to be similar, but, many uses have had various issues attempting to bridge to a Cisco PBX, in the past.

 

[gbrown100]

If you are using a Bridge trunk at the 3CX end, then I'm going to assume it is set as a master, to allow the Avaya to register. When the Avaya registers, it will tell 3Cx the IP it wishes to be "contacted" on.(check a registration to confirm this) As (I assume), the Avaya has a public IP, then that is the one that 3CX will be "communicating" with. As with any remote devices (STUN, SBC) connected to 3CX, the PBX should not be able to contact local IPs, behind a router, directly. It would be using the Public IP plus the port number. Using port forwarding, the remote router uses the port number to identify which local IP, is to be the destination. If port 5060 is currently in use at the Avaya end, then it should be registering with a different port.

I'm not certain if the problems are going to be similar, but, many uses have had various issues attempting to bridge to a Cisco PBX, in the past.

Thanks leejor (as always)

The two systems are on a lan so it makes sense to have them both communicating directly! I have made progress, I have configured the LAN2 port of the Avaya on the same subnet, set the trunk up again and I now have two way communication from 3CX to Avaya phone and when using 3CX to route public calls through the Avaya! The only thing not working is making an internal call from an Avaya phone through to a 3CX extension but again not 3CX's fault here per se, I don't see an attempt from the Avaya to route the call so it's probably an internal config issue on the Avaya. I'll keep the thread alive as I intend to post a how to at the end of it in case any other poor sap has to try to do this!

 

[leejor]

The only thing not working is making an internal call from an Avaya phone through to a 3CX extension but again not 3CX's fault here per se, I don't see an attempt from the Avaya to route the call so it's probably an internal config issue on the Avaya.

I have to assume that is is an Avaya routing issue. Is the Avaya "allowed" to make a non-external type call, over a trunk? In other words, does it expect to be bridged with another PBX?

 

[JohnS_3CX]

This document may be useful to help you understand what 3CX expects to see SIP-wise
https://www.3cx.com/docs/

As for the Avaya routing, that's one you will have to figure out

 

[gbrown100]

*** Update
Well, as per below I have noticed two different behaviours...
1. Ext 355 on Avaya is a SIP client. That is what is giving the results below
2. A Digital extension on the Avaya never spawns a call to 3CX.

I have two issues then. I am giving up for the weekend and will look afresh next week.


OK, so I have pretty much everything worked out bar this:



The Avaya is making the call, extension 4000 exists but 3CX is sending this back! IP Ending 93 is 3cx, ending 4 is avaya. I am 100% sure the trunk password is correct for user 10000 and have reset several times to ensure as such.

From the logs:

08/14/2020 4:31:22 PM - [CM102001]: Authentication failed for AuthFail Recv Req INVITE from 10.91.10.4:5060 tid=92552b7615c25e513b9699c25315fce9 Call-ID=b684522917d592dd2533574c4217b76c: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.91.10.4:5060;rport=5060;branch=z9hG4bK92552b7615c25e513b9699c25315fce9 Max-Forwards: 70 Contact: "10000" <sip:[email protected]:5060;transport=udp> To: <sip:[email protected]> From: "10000" <sip:[email protected]>;tag=b7a73a4ba1ccdb4e Call-ID: b684522917d592dd2533574c4217b76c CSeq: 1076272409 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Proxy-Authorization: Digest username="10000",realm="3CXPhoneSystem",nonce="414d53595f36ae1152:ae3c2ce17f122a49a7d281be467117d6",response="2199149059ce132c80ea6eea09eb5cba",uri="sip:[email protected]" Supported: timer User-Agent: IP Office 11.0.0.2.0 build 23 P-Asserted-Identity: "10000" <sip:[email protected]:5060> P-Preferred-Identity: "10000" <sip:[email protected]:5060> Content-Length: 224 v=0 o=UserA 2496643307 3300750454 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 4:31:22 PM - [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification: Invite-UNK Recv Req INVITE from 10.91.10.4:5060 tid=92552b7615c25e513b9699c25315fce9 Call-ID=b684522917d592dd2533574c4217b76c: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.91.10.4:5060;rport=5060;branch=z9hG4bK92552b7615c25e513b9699c25315fce9 Max-Forwards: 70 Contact: "10000" <sip:[email protected]:5060;transport=udp> To: <sip:[email protected]> From: "10000" <sip:[email protected]>;tag=b7a73a4ba1ccdb4e Call-ID: b684522917d592dd2533574c4217b76c CSeq: 1076272409 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Proxy-Authorization: Digest username="10000",realm="3CXPhoneSystem",nonce="414d53595f36ae1152:ae3c2ce17f122a49a7d281be467117d6",response="2199149059ce132c80ea6eea09eb5cba",uri="sip:[email protected]" Supported: timer User-Agent: IP Office 11.0.0.2.0 build 23 P-Asserted-Identity: "10000" <sip:[email protected]:5060> P-Preferred-Identity: "10000" <sip:[email protected]:5060> Content-Length: 224 v=0 o=UserA 2496643307 3300750454 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 4:31:22 PM - [CM302002]: Authentication failed due to unidentified source of: SipReq: INVITE [email protected] tid=92552b7615c25e513b9699c25315fce9 cseq=1076272409 INVITE [email protected]:5060 / 1076272409 from(wire)