Cisco 7970

[JaseGW]

Hi all

May have been covered before, but confused is an understatement.

So, my problem.....

Had a Cisco connected to a local 3CX server, no problem, calls in and out.
Now I have setup the main company, I decided to run 3CX SBC instead, so as my own phone can connect to the office.
Fantastic, for making calls. Nothng comes in !

Here is the log....

23/03/2021 21:22:19 - Malformed SIP request received:
BadSip Recv 400/INVITE from 127.0.0.1:5080 tid=9f5143271d18a027 Call-ID=yVJtWhmBMxJzlQz9xI3J3A..:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-524287-1---9f5143271d18a027;rport=5060
Record-Route: <sip:[email protected]:5080;user=proxy;uri=sbc.9d0e2dec>
To: <sip:[email protected]>
From: "+44xxxxxxxxxx:IN - 0117 427 xxxx"<sip:[email protected]:5060;nf=e>;tag=7c47421c
Call-ID: yVJtWhmBMxJzlQz9xI3J3A..
CSeq: 1 INVITE
Date: Tue, 23 Mar 2021 21:22:19 GMT
Warning: 399 Bad Request - 'Malformed/Missing Record Route'
Content-Length: 0

Just so you know, the 'x' I added, blanking out my numbers.

 

[jcostlow]

The Cisco 7900 series won't work behind an SBC. They will only work when on the same LAN as the PBX.

https://www.3cx.com/sip-phones/cisco-7940g-7960g/

 

[JaseGW]

Thanks.

I get that I guess...
So bridging two systems is the only option ?
Unfortunately, with the cost per system, this is not an option for the phone at home :-(

 

[JaseGW]

So, my next question is gonna be, whats a good phone to use with SBC, but must have at least 8 lines/accounts ?
hence the Cisco 7970 was a good choice (at the time)

 

[jcostlow]

It is because of how those phones work and now they are EOL.

You can use the webclient or ios/android app in the mean time.

We have used Snom D735 and D785 but the Yealink phones are good too.

At the moment Fanvil seems to be preferred but any phone listed here https://www.3cx.com/sip-phones/ will work.

 

[JaseGW]

Just to complicate things....

What if I had say a Fanvil X4 (or silimar) theyhave only 2 SIP lines, connected one up to 3CX as an extension, but then routed all DID's to it, would there be a way that the phone would know what DID is in use, and light a button say, to indicate this ?

Basically, I have a customer that has 8 tel numbers, one for various companies/projects. He wants to know what line they are calling in on, and wants the ability to select the line he dials out on (for CLI)_

 

[jcostlow]

You could name the inbound rule and the display would show the name of the rule when the call comes in.

To see all DIDs would probably be best to look at the switchboard from the webclient.

For making outbound calls and choosing the number you would make outbound rules and use dial prefixes to designate what number should be sent. The carrier has to allow. Some make you use the primary trunk number only.

 

[dkdocks]

Sorry to bring this thread up from the dead - I have had a similar struggle with 7961G phones @
https://www.3cx.com/community/threa...n-register-in-pcap-has-internal-lan-ip.82078/

Is there any good analysis on what creates/controls the

Warning: 399 Bad Request - 'Malformed/Missing Record Route'

Response from the Cisco on the invite from 3cxSBC vs 3cxLocalInstance - I see the phones show up in 3cx cloud instance, but the source trace is so similar, I'm just wondering if it's a config parameter, firmware 8-5-4(1) vs. 9... issue etc - I mean a sip client is a sip client - should in theory work as a sip client.

I have two ISP's here so in theory the Cisco could get a public IP, no NAT if I could figure out option 66 for it to go direct over public to 3cx cloud....
I tried same with a quick-e asterisk install in AWS and it would fail to register as well not answering the 401 - it's just I find it odd, that if lack of nat advertisement was it - one would think phones local to 3cxSBC would handle INVITE normally yet they go and say 399.

 

[jcostlow]

After reading this post and your other one it looks to be from the phone adding Record-Route in the SIP header.

The phone sends the request and adds Record-Route to the header:

INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-524287-2---9a699811371f4d2b;rport
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-524287-1---9a699811371f4d2b;rport=5060
Max-Forwards: 68
Record-Route: <sip:[email protected]:5060;user=proxy;tnlid=sbc.fc8a8d2d>
Contact: <sip:[email protected]:5060>
To: <sip:[email protected].3cx.us>
From: "My Name 1"<sip:[email protected]:5060>;tag=1dbbef2d
Call-ID: N0JMpmiZTNw5mx6Hkg09dQ..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 402

3CX replies to the invite without Record-Route and the phone doesn't like it. I don't think you will be able to get this working unless the phone quits adding Record-Route.

BUT with the limitations of this phone I don't think it is worth the time messing with it.