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pspenn Brekeke Member
Joined: 28 Oct 2009 Posts: 16
Location: Atlanta, GA
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Posted: Sat Jul 24, 2010 8:42 am Post subject: Thru Registration and call origination issues |
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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
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I appreciate any suggestions.
Thanks,
Perry |
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taitan Brekeke Master Guru
Joined: 15 Mar 2008 Posts: 237
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Posted: Mon Jul 26, 2010 3:40 pm Post subject: |
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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? |
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pspenn Brekeke Member
Joined: 28 Oct 2009 Posts: 16
Location: Atlanta, GA
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Posted: Mon Jul 26, 2010 5:11 pm Post subject: |
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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 |
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taitan Brekeke Master Guru
Joined: 15 Mar 2008 Posts: 237
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Posted: Tue Jul 27, 2010 11:14 am Post subject: |
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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? |
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pspenn Brekeke Member
Joined: 28 Oct 2009 Posts: 16
Location: Atlanta, GA
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Posted: Tue Jul 27, 2010 5:49 pm Post subject: |
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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 |
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taitan Brekeke Master Guru
Joined: 15 Mar 2008 Posts: 237
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Posted: Wed Jul 28, 2010 11:25 am Post subject: |
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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. |
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pspenn Brekeke Member
Joined: 28 Oct 2009 Posts: 16
Location: Atlanta, GA
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Posted: Mon Aug 02, 2010 12:16 pm Post subject: |
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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 |
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taitan Brekeke Master Guru
Joined: 15 Mar 2008 Posts: 237
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Posted: Thu Aug 05, 2010 4:39 pm Post subject: |
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Perry,
it seems you made it.
yeah. using of two SIP servers makes it simpler. |
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