Configuring SIP server to work with AS5300

jaldulaimi · Post by **jaldulaimi** » Sun Mar 30, 2008 8:39 pm

1. Brekeke Product Name and version:SIP 2.1

2. Java version:1.6

3. OS type and the version: Linux Fedora 8

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

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :1 and 2

6. Your problem:

I'm very new to sip and VOIP in general, we just recently bought Brekeke SIP 2.1 standard, I've installed it, it's up and running. I need some advice on how to configure it so SIP-to SIP, phone to sip or sip to phone all work fine. I've already started testing sip to sip calls using Xlite, I had some successes, but I think I'm going to need a lot of help to fully understand dial plan, any advice please knowing the main aim is to make it working with AS5300 and portabilling?

Jassem

james · Post by **james** » Mon Mar 31, 2008 10:13 am

Did you check the Dial Plan Tutorial already?
http://www.brekeke.com/download/download_sip_doc_en.php

You may find lots of samples. (for example, connecting of a gateway..)

jaldulaimi · Post by **jaldulaimi** » Wed Apr 02, 2008 3:01 am

James,

Couple of question;

1-Server configuration, any advise on which parts that I really should not leave to default?
2-Any idea where can I find what type of codec the server is been
set to?

3-Any elaboration on Domains

Your help greatly appreciated.

james · Post by **james** » Wed Apr 02, 2008 10:52 am

Hi

>>1-Server configuration, any advise on which parts that I really should not leave to default?

If you want to use the gateway, you need to add the DialPlan rule.

For example,
-------------------------
[Matching Patterns:]
$request = ^INVITE
To = sip:9(.+)@

[Deploy Patterns:]
To = sip:%1@<GATEWAY_IP_ADDRESS>
-------------------------
By the above DialPlan rule, all calls which have the prefix-9 will be routed to the gateway.

>> 2-Any idea where can I find what type of codec the server is been set to?

The SIP Server doesn't care about codecs. It allow any codecs.

jaldulaimi · Post by **jaldulaimi** » Wed Apr 02, 2008 3:14 pm

James,

many thanks for your time. Is it correct to assume the following;

1-In case of SIP-2-SIP calles, by creating users with passwodrs is good enough to allow only authorized users. No dailup rules are needed.

2-In caes of Sip-2-Phone or Phone-2-SIP, the gateway will decide whether to allow the call or not. Dailup rules is a must so the sip server can route calles.

ALSO, can you please advise when to use and not to use Domains.

Jassem

james · Post by **james** » Thu Apr 03, 2008 1:42 pm

Hi,

>> 1-In case of SIP-2-SIP calles, by creating users with passwodrs is good enough to allow only authorized users. No dailup rules are needed.

Yes.
If both SIP clients are registered in the SIP Server, you don't have to use DialPlan because the SIP Server knows where the SIP client is located based on the registration info.

2-In caes of Sip-2-Phone or Phone-2-SIP, the gateway will decide whether to allow the call or not. Dailup rules is a must so the sip server can route calles.

Yes.
For making a call to a gateway, you need to use DialPlan because a gateway will not register to the SIP Server and the SIP Server doesn't know where the gateway is located.

>> ALSO, can you please advise when to use and not to use Domains.

What does your "Domain" mean?
Is it Multiple-Domain feature?

jaldulaimi · Post by **jaldulaimi** » Thu Apr 03, 2008 2:50 pm

James,

Again, many thanks for your time.

About Domains, I'm just trying to figure out on how to use them, are we talking about FQDN here?

If you just explain to me a bit when to use them, and what's the idea behind using multiple domains? Please give me some examples.

Jassem

james · Post by **james** » Fri Apr 04, 2008 10:32 am

Hi

>> are we talking about FQDN here?

Yes. I mean FQDN.

>> what's the idea behind using multiple domains?

With the Multiple-Domains feature, the SIP Server can host multiple FQDN. So you can deploy one SIP server for multiple services.

jaldulaimi · Post by **jaldulaimi** » Tue Dec 02, 2008 4:33 pm

Hi Guys,

I hope someone can help.

Let's assume, I've got the following dial plan;

[Matching Patterns:]
$request = ^INVITE
To = sip:0011(.+)@

[Deploy Patterns:]
To = sip:74763%1@IP of the gateway

On AS5300, I will need two dial peers, one VOIP to receive those calls, and another POTS to terminate them. I've been told I get to include a prefix and make AS5300 to strip and dial only the required number. Example, according to my sip dial plane, when a user dial, 00116175695695, what will be sent to AS5300 is 7476300116175695965, my dial-peer should strip the 74763 and send only 00116175695695 to the PSTN. My problem not sure on how to do this, can someone please give me an example on how to do it, striping the prefix?

Your greatly appreciated.

Jassem

hope · Post by **hope** » Fri Dec 05, 2008 1:29 pm

[Matching Patterns:]
$request = ^INVITE
To = sip:0011(.+)@

[Deploy Patterns:]
To = sip:74763%1@IP of the gateway

With above dial plan, when a user dial, 00116175695695, what will be sent to AS5300 is 747636175695965, not 7476300116175695965

if send 7476300116175695965 to GW

[Matching Patterns:]
$request = ^INVITE
To = sip:(0011.+)@

[Deploy Patterns:]
To = sip:74763%1@IP of the gateway

jaldulaimi · Post by **jaldulaimi** » Sat Dec 06, 2008 9:53 pm

Hi,

Somehow, this dial plan works for me and I do make a call, but only if I get rid of my TCL script, do you know anything about TCL and AS5300?

Regards,

Jassem

Configuring SIP server to work with AS5300

Configuring SIP server to work with AS5300

AS5300 and Brekeke Sip server