ARS rule for unregistered clients

svrtelecom · Post by **svrtelecom** » Fri Jul 01, 2011 10:05 am

1. Brekeke Product Name and version:
Brekeke PBX V2.1.5.6 & V2.4.9.0

2. Java version:
Java 6 Update 25

3. OS type and the version:
Windows XP 32 bit patched up to date

4. UA (phone), gateway or other hardware/software involved:
Brekeke PBX

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
All on same LAN

6. Your problem:
We have a number of PBXes at different site (all Brekeke PBX). Each site has distinct extension numbers, e.g 2xx on one site, 3xx on another. We have ARS rules so that just dialling any extension number connects to the relevant sites.

We have been using the older version V2.1.5.6 and want to upgrade to V2.4.9.0 (the latest version. Now our inter-PBX calls don't work. We have looked at the wiki article "Update Issues from v2.2.7.7 and earlier versions" and tried to apply those modifications, but still they won't connect.

We have added the "IN" pattern to the ARS, as per the wiki article but are not clear what the deploy pattern "To:PBX_user" means. Is this typed in exactly as that, or is it a placeholder for something else? We have

MATCHING PATTERN
From sip:(.+)@192.168.2.199
To sip:(.+)@

DEPLOY PATTERN
To PBX_user

We have tried various other things in the Deploy:To field, like a PBX user name, but we can't get it to accept the incoming calls.

We have also added a dial plan to the SIP server (Brekeke's integral SIP server)

MATCHING PATTERN
$request=^INVITE
$addr=192.168.2.199 (address of other PBX for incoming call)

DEPLOY PATTERN
$auth=false
$continue=true

Calls from the newer version PBX to the older one work fine (as you would expect) but we can't get calls from the old to the new PBX to connect.

Thanks in anticipation

hope · Post by **hope** » Fri Jul 01, 2011 2:57 pm

use the second dial plan in http://wiki.brekeke.com/wiki/Update-issues-post-v2277

svrtelecom · Post by **svrtelecom** » Wed Jul 13, 2011 11:32 am

We have tried the second SIP server dial plan - no change, still doesn't work.

The called PBX replies "486 Busy Here" and the call fails. Calls from the new PBX to the old version (2.1.5.6) work fine.

I'm not sure what should be typed in the ARS "IN" pattern. The deploy pattern "To:" says PBX_user, but what goes in PBX_user? That string literally pasted in there doesn't work. Do we need a specific user?

Thanks in anticipation for any help

hope · Post by **hope** » Wed Jul 13, 2011 12:16 pm

yes, in ARS pattern-IN deploy pattern To, need to specify a pbx user number which will answer the incoming call such as a pbx user number of auto attendant

svrtelecom · Post by **svrtelecom** » Mon Jul 18, 2011 11:34 am

Thanks Hope

That doesn't quite do what we want. Sending all the calls from a VoIP device to one number defeats the object.

For example, we have an ISDN gateway (Patton SN4960) which accepts incoming calls from PSTN. There is a number range associated with this, to give a DDI (Direct Dial In) facility, so according to the National Number dialled on the PSTN, the call gets routed to various internal extensions.

The incoming Caller ID means that the call is shown as being "From" whatever National Number originated the call. Now, I am assuming that the PBX won't accept a call "From" an unregistered user ID - that being the security update that stops our happily working system from being upgraded, I assume...

Is there a way to allow the PBX to accept calls coming from a particular IP address (for example) regardless of whether the caller is authenticated?

Thanks again for taking the trouble to reply.

hope · Post by **hope** » Mon Jul 18, 2011 12:51 pm

so according to the National Number dialled on the PSTN, the call gets routed to various internal extensions.

So the To field sending from gateway is already the number existing as a pbx user in pbx?
if yes,
1. add dial plan at sip server side on top of all other dial plan to accept calls from gateway patton sn 4960
need to click "Apply Rules" button after adding dial plan
[Matching Patterns]
$request = ^INVITE
$addr = patton_gateway_IP
To = sip:(.+)@

[Deploy Patterns]
$auth = false
To = sip:%1@127.0.0.1:15060

2. at pbx ARS rule set as following
[Matching Patterns]
From:sip:(.+)@patton_gw_IP
To: sip:(.+)@

[Deploy Patterns]
To: $1

$1 means the content in parenthesis in matching pattern To

if call is not sent to related pbx user, capture packets at brekeke server pc to see what From and To header is sent from gateway to brekeke and change ARS rule to the same as what sent from gateway

svrtelecom · Post by **svrtelecom** » Wed Jul 20, 2011 11:54 am

thanks again Hope

Still no positive result. We have run Wireshark, and looked at the SIP packets. The incoming call is offered (INVITE) to 1000 - which is a valid user / extension on the PBX. The PBX replies "486 busy here" and the caller gets busy tone...

Is there some other option we can try? Or some more diagnostic work we can do?

We would really like to upgrade to the new version of Brekeke, but this is an important feature for us, and unless we can solve this, we are stuck with V2.1.5.6

Thanks again for taking the trouble.

Regards

hope · Post by **hope** » Wed Jul 20, 2011 2:48 pm

change ARS rule pattern-IN as following and set its priority with larger number than all other ARS rules pattern-IN

[Matching Patterns]
From:leave it blank
To: sip:(.+)@

[Deploy Patterns]
To: $1

if works, check invite packets about the calls sent from gateway, what is after @ in INVITE From header sip:xxx@xxx
and set it in above ARS rule From after @ as sip:.+@what_in_packets

svrtelecom · Post by **svrtelecom** » Wed Jul 27, 2011 11:11 am

Thanks a lot, Hope

We have tested both our scenarios using the July 18 dial plan and the above ARS IN rule, and it works fine. So we are now free to upgrade our PBXes.

Thanks again for taking the time to help out.

hope · Post by **hope** » Wed Jul 27, 2011 2:54 pm

no problem, but it is better to change ARS pattern-IN to specify more specific From and To in the matching pattern to only match calls from gateway
otherwise, all calls(even calls between pbx extensions) through pbx will apply to this pattern-IN

the changes depend on the INVITE sent from Gateway