Qualify Vega 50 BRI with asterisk > 1.4

FreePBX Installation / Upgrade
1

Hello !

I have a Vega 50 BRI sip gateway what is responding ok at qualifying with asterisk 1.4.

When i install any other version of fresh FreePBX (asterisk 1.8 & uppers), this qualify failed with UNREACHABLE message. I install another fresh install in a virtual machine with asterisk 1.4, and the response is ok.

The asterisk server & gateway are in the same network/switch

This is the sip dialog with asterisk 1.4 at qualifying (OK response):

Reliably Transmitting (no NAT) to 192.168.0.10:5060:
OPTIONS sip:192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1da1ac1a;rport
From: “Unknown” sip:[email protected];tag=as2c1d6fce
To: sip:192.168.0.10
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 21 Oct 2014 20:38:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

elastix*CLI>
<— SIP read from 192.168.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1da1ac1a;rport
From: “Unknown” sip:[email protected];tag=as2c1d6fce
To: sip:192.168.0.10;tag=0012-0167-47E5B035
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: sip:192.168.0.10:5060;maddr=192.168.0.10
Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER,PRACK
Supported: 100rel, replaces
Accept: application/sdp, application/none
Accept-Language: en
User-Agent: VEGABRIS/06.02.06xS019
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS

######################################################################################################

This is the sip dialog with asterisk 1.6, 1.8, 11, 12 or 13beta: (UNREACHABLE)

Reliably Transmitting (no NAT) to 192.168.0.10:5060:
OPTIONS sip:192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK73f201b4
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as2d56b434
To: sip:192.168.0.10
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.1)
Date: Tue, 21 Oct 2014 15:23:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK73f201b4
From: “Unknown” sip:[email protected];tag=as2d56b434
To: sip:192.168.0.10;tag=0013-044E-4A394683
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: sip:192.168.0.10:5060;maddr=192.168.0.10
Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER,PRACK
Supported: 100rel, replaces
Accept: application/sdp, application/none
Accept-Language: en
User-Agent: VEGABRIS/06.02.06xS019
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Retransmitting #1 (no NAT) to 192.168.0.10:5060:
OPTIONS sip:192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK73f201b4
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as2d56b434
To: sip:192.168.0.10
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.1)
Date: Tue, 21 Oct 2014 15:23:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK73f201b4
From: “Unknown” sip:[email protected];tag=as2d56b434
To: sip:192.168.0.10;tag=0013-044E-4A394683
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: sip:192.168.0.10:5060;maddr=192.168.0.10
Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER,PRACK
Supported: 100rel, replaces
Accept: application/sdp, application/none
Accept-Language: en
User-Agent: VEGABRIS/06.02.06xS019
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Retransmitting #2 (no NAT) to 192.168.0.10:5060:
OPTIONS sip:192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK73f201b4
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as2d56b434
To: sip:192.168.0.10
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.1)
Date: Tue, 21 Oct 2014 15:23:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK73f201b4
From: “Unknown” sip:[email protected];tag=as2d56b434
To: sip:192.168.0.10;tag=0013-044E-4A394683
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: sip:192.168.0.10:5060;maddr=192.168.0.10
Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER,PRACK
Supported: 100rel, replaces
Accept: application/sdp, application/none
Accept-Language: en
User-Agent: VEGABRIS/06.02.06xS019
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Retransmitting #3 (no NAT) to 192.168.0.10:5060:
OPTIONS sip:192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK73f201b4
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as2d56b434
To: sip:192.168.0.10
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.1)
Date: Tue, 21 Oct 2014 15:23:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK73f201b4
From: “Unknown” sip:[email protected];tag=as2d56b434
To: sip:192.168.0.10;tag=0013-044E-4A394683
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: sip:192.168.0.10:5060;maddr=192.168.0.10
Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER,PRACK
Supported: 100rel, replaces
Accept: application/sdp, application/none
Accept-Language: en
User-Agent: VEGABRIS/06.02.06xS019
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Retransmitting #4 (no NAT) to 192.168.0.10:5060:
OPTIONS sip:192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK73f201b4
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as2d56b434
To: sip:192.168.0.10
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.1)
Date: Tue, 21 Oct 2014 15:23:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK73f201b4
From: “Unknown” sip:[email protected];tag=as2d56b434
To: sip:192.168.0.10;tag=0013-044E-4A394683
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: sip:192.168.0.10:5060;maddr=192.168.0.10
Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER,PRACK
Supported: 100rel, replaces
Accept: application/sdp, application/none
Accept-Language: en
User-Agent: VEGABRIS/06.02.06xS019
Content-Length: 0

<------------->
— (13 headers 0 lines) —

###################################################################################

Can anyone understand this and help me ?

Thank’s for all,
Aitor.

2

Hi - This is not the Elastix or the Asterisk forum. I doubt this is a FreePBX issue. I would however suggest you post your trunk settings. Many things changed between 1.4 and 1.8 WRT SIP peer syntax.

Have you reviewed the Asterisk sip.conf sample to understand what the trunk settings do?

3

Aupa,

yes, you’re right, but my initial install is over elastix with asterisk 1.4, which has been running smoothly for a few years, and so I got to meet the wonderful freebpx.

this is the simplest configuration that I used for testing in trunk peer details:

host=192.168.0.10
type=peer
qualify=yes

the gateway is accepting guest & not validated users (inside local network) and I have not changed anything in the configuration (in new installation of version 1.4 the qualify is OK).

Yes, I have tried many configuration options trunk and many hours of testing I’ve been reading in the forums. This simple configuration for testing is the result of that.

best regards,
berrigorri.

4

If that trunk is local and trusted then maybe

host=192.168.0.10
type=friend

is all you need after you setup the vega correctly. Reference

http://wiki.sangoma.com/Vega-50-Technical-Documentation

5

but this is the same as qualify=no —> Unmonitored

and I lose a lot of functionality. For example even hang up the call, the tone still sounds in the target phone. And I lose many more features.

thank’s !!

6

If your machines are on the same network then qualifying it is almost certainly a waste of time and bandwidth, if one can’t see the other (either way) for SIP traffic then you have to go back and check your setup, it’s probably not FreePBX.

7

ping is <1ms

and if I install the 1.4, then qualify response is OK. I am sure that is not the freepbx, but a problem of asterisk.

any idea what may have changed after version 1.4 of Asterisk about qualifying?

8

Again you should read the source documentation:-