Solved Spent the night fighting with a Cisco 7961G to 3cx hosted - 407 on REGISTER in pcap - has internal lan ip

[dkdocks]

This is what works:
IP of cloud instance changed to Exention 8000 running on 3xc at IP 123.45.56.789 (fake)
Local lan behind nat is 192.168.0.0/24 (fake)

1. I can SIP Register a soft client SIP Linphone with STUN/ICE disabled from same local network to 3cx hosted instance.
as To: [email protected] From: [email protected] ... 401 please authenticate - sends credentials from "Authentication Tab" from 3cx.

2. I've convinced the 7961G (Firmware SIP41.8-5-4S) to SIP Register with Asterisk running locally
sip.conf excerpt:

[8000]
type=friend
host=dynamic
nat=no
username=8000
secret=password
context=home
qualify=yes

SEP123456789000.cnf.xml (MAC:123456789000)
excerpt I'm fighting with based on pcap from the 3cx hosted/cloud;

REGISTER sip:123.45.56.789 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9dd2279356
From: <sip:[email protected]>;tag=0016ce020e00151e893b60-5f072e0e
To: <sip:[email protected]>
Call-ID: 0016c70e-03cf00fe-3fbc1ee6-843fc0e1@192.168.0.25
Max-Forwards: 70
Date: Wed, 16 Dec 2009 08:19:13 GMT
CSeq: 120 REGISTER
User-Agent: Cisco-CP7961G-GE/8.5.3
Contact: <sip:100@192.168.0.25:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00000000000>";+u.sip!model.ccm.cisco.com="308"
Supported: (null),X-cisco-xsi-7.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:25 Name=SEP123456789000 Load=SIP41.8-5-4S Last=initialized"
Expires: 3600

Where .cnf.xml contains

<natEnabled>true</natEnabled>
<natAddress>789.10.11.12</natAddress>

#my isp external ip
also tried variations of

<proxy>123.45.56.789</proxy> vs. 
<proxy>USECALLMANAGER</proxy>

Phone stays with spinny wheel "Registering" and keeps resending register requests as per PCAP.

Nothing gets 7961G in pcap done from the remote 3cx to say anything but 192.168.0.25:5060 instead of 789.10.11.12...

3CX does not return 401 but instead just keeps returning 407 - The button for enable direct sip is enabled for EXTN, (rather this is turned off - Disallow use of extension outside the LAN (Remote extensions using Direct SIP or STUN will be blocked)

I've poured over the forum related to registering 79x1 phones, but the threads are dated, and some connect to local 3CX and with lower version numbers than 16,
this one may behave differently, and the auto- .cnf.xml feature or phone itself is not listed in the dropdown to provision in 3CX hosted, however since it takes a software SIP client
in theory it should work.

Next i'll attempt to SIP register this 7961G with asterisk running in AWS
if anyone with any ideas I'm welcome to hear them please.

 

[dkdocks]

Solved issue by installing SBC on the same raspberry pi that ran asterisk locally - phone 7961 Cisco SIP registered with SBC and SBC is connected to cloud instance. I could not get it to work with remote ip of asterisk 16 pbx test instance in aws either.

 

[leejor]

It's been a few years since I set up a few Cisco 79XX sets (I think it was a 7960, a 7961 and a 7941). I don't recall them supporting STUN (it is unaware of it's public IP is), which may cause an issue, if the set is not on the same LAN, as the server, or you don't use an SBC, or VPN.

 

[dkdocks]

It's been a few years since I set up a few Cisco 79XX sets (I think it was a 7960, a 7961 and a 7941). I don't recall them supporting STUN (it is unaware of it's public IP is), which may cause an issue, if the set is not on the same LAN, as the server, or you don't use an SBC, or VPN.

I set up an 3xcsbc 16 on Raspberry Pi on a subnet next to the phones - it works now with the 3CX hosted by 3CX and 7961 phones registered, phones showing up in the phones list - but their acting weird, 408 Timeout - and etc.. calls going directly to voicemail. I'll guess I'll have to tcpdump/wireshark it. Or debug the xml config a bit more.
Local CUCM machine was killing too much power, needed asterisk or it's own border gw on to send calls out Twilio and Voicepulse it was all a bit much- but the phone's I wanted to recycle with 3CX.

In theory the phones should have obeyed the directive in the cnf, and in sip pcap I saw them do this but for some reason they would not respond to 407 nor 401 must auth digests to any cloud sip servers i'd set up.
<natEnabled>true</natEnabled>
<natAddress>mypublic-ip</natAddress>

Thanks for your reply.

 

[dkdocks]

Nah not so lucky, not working - 3cxsbc v16 + 7961 phone - no joy - 3cx cloud system sees the 7961 phones but they can't even call each other, you can listen to voicemail greetings and thats about as far as I got. This is after assuming that 3cxsbc and 7961 should be on same vlan, cracking into my 3750 and stripping the voip vlan configs out from poe ports.

There is probably no method to make 7961/7941 behave sane or I'm missing some config setting in xml

Log from 3xc SBC is as follows (redacted local lan and 3cx fqdn)

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]>
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


v=0
o=3cxPS 32303274774233088 25503977395191809 IN IP4 127.1.2.3
s=3cxPS Audio call
c=IN IP4 192.168.1.50
t=0 0
m=audio 20002 RTP/AVP 0 8 9 18 112 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 opus/48000/2
a=fmtp:112 maxplaybackrate=16000;sprop-maxcapturerate=16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
sigcomp id=
DEBUG | 20210612-202407.767 | 3CX | RESIP:TRANSPORT | 1969800048 | Transport.cxx:392 | incoming from:
[ V4 192.168.1.178:49196 UDP flowKey=12 transportKey=1 ]
STACK | 20210612-202407.768 | 3CX | RESIP:TRANSPORT | 1969800048 | Transport.cxx:393 |


SIP/2.0 400 Bad Request
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
To: <sip:[email protected]>
From: "My Name"<sip:[email protected]:5060>;tag=1dbbef2d
Call-ID: N0JMpmiZTNw5mx6Hkg09dQ..
CSeq: 1 INVITE
Date: Wed, 16 Dec 2009 08:15:12 GMT
Warning: 399 Bad Request - 'Malformed/Missing Record Route'
Content-Length: 0

STACK | 20210612-202407.768 | 3CX | RESIP:TRANSACTION | 1969800048 | TransactionState.cxx:735
| Found matching transaction for SipResp: 400 tid=9a699811371f4d2b cseq=1 INVITE / 1 from(wire) -> tid=9a699811371f4d2b
[ ClientInvite/Calling unreliable target=[ V4 0.0.0.0:0 UNKNOWN_TRANSPORT ]]

Giving up as I tried a D70, works flawlessly via manual config without need for 3cxSBC, time to retire the Cisco's.

 

[JohnS_3CX]

Hi,

Those phones are not supported on our hosted platform or as remote phones. Read the limitations here:
https://www.3cx.com/sip-phones/cisco-7940g-7960g/#h.xxaqfkri8bzy

You are indeed better off retiring them and replacing them with something from our supported phones list:
https://www.3cx.com/sip-phones/