strip leading characters from caller id

1. Brekeke Product Name and version: BSS 2.3.6.0

2. Java version: build 1.6.0_07-b06

3. OS type and the version: Win XP

4. UA (phone), gateway or other hardware/software involved: Cisco Call Manager / 7921 + rauland nurse call

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

6. Your problem:
I need to strip leading characters from caller id display. the important characters get cut off by the limited caller id size. I have been able to replace but not shoerten.

ex.
current display
From: 10th 1025 1 Patient<sip:1025*1@192.168.6.201:5060>

need display
From: Patient<sip:1025*1@192.168.6.201:5060>

current dial plan

Matching:
$request=^INVITE
To=sip:(.+)@

Deploy:
to=sip:%1@10.200.2.9

matching patterns:
$request=^INVITE
From=".+ Patient"(.+)
To=sip:(.+)@

deploy patterns:
From="Patient"%1
To=sip:%2@10.200.2.9

I tried your solution and a few variations, for some reason I no longer geerate a call at all if I have From= in the deploy pattern. My application is not phone to phone or pc to phone it is a speaker console in a room that is connectected to a gateway server which registers the rooms to BSS. I am starting to wonder if the gateway requires that there be no changes to the header so it can find the destination room?

how about
matching patterns:
$request=^INVITE
From=".+ Patient"(.+) ;.+
To=sip:(.+)@

deploy patterns:
From="Patient"%1
To=sip:%2@10.200.2.9

hope wrote:how about
matching patterns:
$request=^INVITE
From=".+ Patient"(.+) ;.+
To=sip:(.+)@

deploy patterns:
From="Patient"%1
To=sip:%2@10.200.2.9

Does not work either, calls fail to complete, thanks for your efforts.

Hi all,

In my environment, the above DialPlan rule works...

tman,
did you get any error response from UAS?
can you paste an INVITE packet which the SIP Server sent?

janP wrote:Hi all,

In my environment, the above DialPlan rule works...

tman,
did you get any error response from UAS?
can you paste an INVITE packet which the SIP Server sent?

Thanks for all the help, got me on the right path, I did not even know what SIP was until about a month ago. between what you told me and the bss_tutorial dial plan doc I came up with this

$request=^INVITE
From=(.{12}(.+))<(.+@)>
To=sip:(.+)@


From=%2<%3>
To=sip:%4@10.200.2.9

and it works for the most part, I will need tweak it some how because differnt areas have different numbers of lading characters but I think I am ok now.

tman

If your above DialPlan rule works, I'm glad.

TMAN - I am new as well using Brekeke with Responder Gateway server. I am struggling right now to get the Patient Stations to register with Brekeke....any suggestions? If i point the Registrar IP to the Nursecall NIC on the RGS and the Proxy IP to the Hospital LAN IP i get an "unauthenticate" response. if i set the Proxy to the Hospital LAN NIC my RGS just Fails on registration. Any help with RGS and Brekeke setup between the two would be much appreciated

brad051380,

Refer to this Wiki topic.
http://wiki.brekeke.com/wiki/a59dbbb4
It will help you.

got mine to work. It was a port conflict between the Rauland software and Brekeke. I was able to change the port on the Rauland system and it worked fine. Also, i had to force RTP = ON as well to get the Voice Paths to work. THanks for your help. Also, is there a guide with the different variables and what they mean? such as %1,%2 etc.. maybe i am just too impatient to figure it out on my own!lol
Thanks,
Brad

%1, %2 refer to the content in the parenthesis in the matching patterns

%1 refers to the first parenthesis included content in the matching patterns
%2 refers to the second parenthesis included content in the matching patterns