caller number identification

[rhasche]

1. Brekeke Product Name and version:2.4.9.0

2. Java version:

3. OS type and the version:Linux

4. UA (phone), gateway or other hardware/software involved:Quintum DX

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :

6. Your problem:

I want to identify the caller from pstn (quintum DX) on the extension voip phones, but I setup the Quintum to send the caller number the pbx do not accept the call.
EX
From: 34561234@192.168.10.241 to 9010@192.168.10.240

34561234 is caller number and 9010 is the extension
Quintum ip: 192.168.10.241
Brekeke PBX IP: 192.168.10.240

I tried the fowling settings on brekeke but it doesn’t work

ARS Pattern in:
[Matching Patterns]
From: sip:(.+)@192.168.10.241
To: sip:(.+)@

[Deploy Patterns]
To: sip:$2@

Matching Patterns

$request = ^INVITE $addr = 192\.168\.10\.241 To = sip:(.+)@

Deploy Patterns

$auth = false $continue = true To = sip:%1@127.0.0.1:15060

What can i do to fix it?

[hope]

ARS Pattern in:
[Matching Patterns]
From: sip:(.+)@192.168.10.241
To: sip:(.+)@

[Deploy Patterns]
To: sip:$2@ here is $1 which is the first parenthese in [matching pattern] To field and no sip: and @ need

is still not working
try removing setting in [Matching patterns] From field.

[rhasche]

Thanks, I tried it, but the problem still the same.

When we setup the quintum DX to change the caling number for 3000 (quintum user) the call is completed.
We can see at quintum call log for a call from 1933181085 PSTN number to 9070 extension:

CalingNumber CalledNumber CallStartTimer Time Code Incoming/Orig Outgoing/Term
3000 9070 20120419221100 28 1 1 192.168.10.240
The extension 9070 see at the phone 3000 as a caller instead 1933181085


if we don’t change caling number the call is not completed (isdn error 111)

CalingNumber CalledNumber CallStartTimer Time Code Incoming/Orig Outgoing/Term
1933181085 9070 20120419221223 0 111 1 1 192.168.10.240

[hope]

how about set as wiki page below?
http://wiki.brekeke.com/wiki/Quintum-Tenor-DX

if donot change calling number,
can the call be sent out to Brekeke server?
if capture packets at brekeke server can you see the call from gateway?

[rhasche]

the quintum is set like as wiki. When it is sending caller id brekeke do not complete the connection, but when we change the caller id for the quintum UA (3000) it's work well.

Active Sessions
Session ID: 463827
From: sip:3000@192.168.10.240 (192.168.10.241:5060)
To: sip:9070@192.168.10.240 (127.0.0.1:15060)
Time: 2012-04-23 23:00:06.963
Status: Ringing


Session ID: 463829
From: sip:3000@192.168.10.240:5060 (127.0.0.1:15062)
To: sip:9070@192.168.10.240:5060 (192.168.3.58:58751)
Time: 2012-04-23 23:00:06.968
Status: Ringing

brekekke is not accepting calls from not registred clients

[hope]

if put the dial plan rule on top of all other dial plan rules,
can pbx/sip server accept gateway calls?
is source IP of the gateway call from IP 192.168.10.241?

Matching Patterns

$request = ^INVITE $addr = 192\.168\.10\.241 To = sip:(.+)@

Deploy Patterns

$auth = false To = sip:%1@127.0.0.1:15060