Brekeke Forum Index » Brekeke SIP Server Forum

Post new topic   Reply to topic
Thru Registration and call origination issues
Author Message
pspenn
Brekeke Member


Joined: 28 Oct 2009
Posts: 16
Location: Atlanta, GA

PostPosted: Sat Jul 24, 2010 8:42 am    Post subject: Thru Registration and call origination issues Reply with quote

1. Brekeke Product Name and version: 2.4.5.5

2. Java version: 1.6.0.20

3. OS type and the version: Red Hat 5.5

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

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

6. Your problem:

Hello All,
My setup is with BSS on the edge of the network. 2 NIC's on the Internet and 1 NIC on the inside. I would like to register phones from the internet thru to an Asterisk FreePBX server on the inside. All of this works fine. However, when I try to make a call from one of the externally connected phones the call is not originating from Asterisk. Instead it is hitting a matching pattern on BSS which will take it where it needs to go, but the Asterisk server is left out and I want all station calls coming from Asterisk for reporting.

How can I register a phone thru BSS so that all calls from that station will originate from Asterisk?

Here is my Matching Pattern on Registration:
Code:

[MATCHING PATTERN]
$request = ^REGISTER
From=sip:1099912@

[DEPLOY PATTERN]
&net.registrar.thru.allow=true
$action=register


I appreciate any suggestions.
Thanks,
Perry
Back to top
View user's profile
taitan
Brekeke Master Guru


Joined: 15 Mar 2008
Posts: 237

PostPosted: Mon Jul 26, 2010 3:40 pm    Post subject: Reply with quote

I can not understand what you want to do..

Do you want to make a registration from outside phones to the inside Asterisk thru the Brekeke SIP Server?
Back to top
View user's profile
pspenn
Brekeke Member


Joined: 28 Oct 2009
Posts: 16
Location: Atlanta, GA

PostPosted: Mon Jul 26, 2010 5:11 pm    Post subject: Reply with quote

That is correct... Which that is working just fine...
But what I have an issue with is placing a call from the external phone and that call originate from the Asterisk.
I will give you an example:

I register extension 1099912 at my house through BSS to Asterisk.
I want to call an extension on the Avaya server which has a dial rule in BSS, but I want the call to come from Asterisk for reporting. Instead of the call going through BSS to the Asterisk, the call from the external phone takes the matching rule in BSS to the Avaya thus by-passing the Asterisk.

I am looking to use BSS as a Border Controller.

I think I might need to use one Proxy for External and one proxy for Internal. I am not even sure I am making sense anymore... I have been racking my brain all day with this.
Thanks,
Perry
Back to top
View user's profile
taitan
Brekeke Master Guru


Joined: 15 Mar 2008
Posts: 237

PostPosted: Tue Jul 27, 2010 11:14 am    Post subject: Reply with quote

it seems the call should be..
Your extension --> BSS --> Asterisk --> Avaya --> Extension
Is it correct?

With DialPlan, it will be possible.
Can you paste your current DialPlan rules here?
Back to top
View user's profile
pspenn
Brekeke Member


Joined: 28 Oct 2009
Posts: 16
Location: Atlanta, GA

PostPosted: Tue Jul 27, 2010 5:49 pm    Post subject: Reply with quote

I think I see what my problem is and I am not convinced it will be possible with a single Proxy.
I am trying to do this:

Extension --> BSS --> Asterisk --> BSS --> Avaya --> Extension

I believe I need one instance of BSS for the external connections (ITSP and externally connected stations) and then another instance of BSS for internal connections.

My goal in all of this is to keep one single dial plan in a central Proxy and just point all of my other servers to it (Avaya, Genesys SIP Server, Multiple Asterisk Servers, etc...).

Does this sound like it is a normal use of BSS?
Thanks,
Perry
Back to top
View user's profile
taitan
Brekeke Master Guru


Joined: 15 Mar 2008
Posts: 237

PostPosted: Wed Jul 28, 2010 11:25 am    Post subject: Reply with quote

Hi Perry,

Yeah. it will be easy if you use two SIP Servers..
But you can do same thing by a single SIP Server with DialPlan rules..

Try "$addr" variable in DialPlan to check if the sender is Asterisk.

If you paste your rules, I can modify them.
Back to top
View user's profile
pspenn
Brekeke Member


Joined: 28 Oct 2009
Posts: 16
Location: Atlanta, GA

PostPosted: Mon Aug 02, 2010 12:16 pm    Post subject: Reply with quote

Hi Titan,
Sorry for the late reply... I think I got it...
The original request was coming from user@ and was pointing to 4045551212@external_ip.com.
I needed to target the call to the asterisk server and then change both the 'to' and 'from' so they are user@asteriskserver.local and 4045551212@asteriskserver.local.
Asterisk will then process the call as if it were originating from Asterisk. Then when asterisk sends the request back to the proxy, the first rule on the extension no longer applies because it is not coming from the original uri so therefore the call proceeds to the next matching pattern. I hope that just made sense.

I tested this and got two way audio and the call proceeded in to the proxy from the enpoint, then to asterisk, then back to the proxy, then out to the ITSP.
I still believe that I am going to use a second proxy for this because it will keep life and troubleshooting simple and all of my eggs aren't in one basket should I have a problem.

Here is the rule I came up with. I am going to tweak it so that it will match on any endpoint that is registered to the proxy.

Code:

MATCHING:
$request=^INVITE
From=sip:(4201)@
To=sip:(.+)@63.xxx.xxx.xxx <external IP of the Proxy>

DEPLOY:
$target=10.43.128.11
From=sip:%1@10.43.128.11
To=sip:%2@10.43.128.11
$replaceuri=True


Thanks again for the suggestions.
Perry
Back to top
View user's profile
taitan
Brekeke Master Guru


Joined: 15 Mar 2008
Posts: 237

PostPosted: Thu Aug 05, 2010 4:39 pm    Post subject: Reply with quote

Perry,
it seems you made it.
yeah. using of two SIP servers makes it simpler.
Back to top
View user's profile
Display posts from previous:   
Post new topic   Reply to topic    Brekeke Forum Index » Brekeke SIP Server Forum All times are GMT - 7 Hours
Page 1 of 1