Dial Plan and ARS

[hope]

1. Brekeke Product Name and version:

2. Java version:

3. OS type and the version:

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

5. Select your network pattern from

6. Your problem:

It seems like DialPlan and ARS has the same syntax. Do they have the same function? If yes, why do we need both of them?

Thank you

[voipwell.com]

Hi,

The syntax is slightly different. If you have the pbx and sip server most of the configurations are done thru the pbx ARS. If you only have the sip server the configuration is done thru the dial plan. When you have both pbx and sip server dial plan comes with specific entries to work with the pbx and usually never need be changed except for special circumstances.

[colin2710]

Hi Nik,

That's very interesting - we have the PBX with SIP server (2.1), does that mean that if you want to use ARS's then you have to create a user in the PBX. The reason I ask is I am trying to get my head around dial plans/ARS's and I have assumed that if you have PBX users you use ARS's - if you don't (just have SIP users) then you use dial plans. The reason is that we will have/need SIP users, so I have been trying to use SIP server dial plans.

You can what I am trying to do at the moment - below

See: http://www.brekeke-sip.com/bbs/viewtopic.php?t=5662

Regards - Colin

[voipwell.com]

Colin,

Please don't be upset with Brekeke. For years they supported free on this forum while they were developing the product. At this point it has matured into a very stable reliable product that costs a fraction of the other commercially available products with similar features. I think their business model is now focused on developing resellers, "boots on the ground" to support end users for free as well as the many competent users that lurk here or use their paid support options. Fortunately, many of us that have benefited from their support for the last few years feel obligated to help you as they helped us. And we are glad to do it.

If you want to get a feel for this company, read thru the old postings where they were supporting end users. You won't find a kinder more helpful attitude anywhere.

Ok, so back to your specific problem. One of the hardest things to get when starting with Brekeke is the big picture. While the docs address the fine details pretty well, you need a big picture view of how to set things up. Let me try to give you that here and shame on me for not having documented this on my website.

Dialplans vs ARS.
When you purchase the sip server alone the dial plans are empty. when you purchase it with the pbx, there are several dial plans to help the sip server work with the pbx. Common changes to the dial plans that include a pbx would be to have fax calls not use rtp relay, to pass on sip info packets that are not dtmf, like hook flash, to route video calls around the pbx and to add the radius plug in entries for accounting. Things like that.

If you have the Bekeke pbx and sip server you can have a sip user make a call using the ARS even if they are not set up as a pbx user. However, I believe it does count as a call session which is limited by the number of users you have purchased the pbx for. For instance pbx basic is 20 user 10 simultaneous calls of which sip users not set up in the pbx would count. So you can make 10 simutaneous calls thru the ARS by pbx and non pbx users. The pro version(20 user) allows 20. You need the ARS to place calls to providers that require authentication which is not ip address based. Although with upper registration I believe it can be done but since I spend all my time on the pbx I havn't used it. You might check it out though.

Ok, so your next question should be "Then why set up a pbx user"?
Good question.

Well, when a call comes in thru the ARS the destination is specified as a user. When that call flows to a user it all the great pbx features of how to handle the call is in the user screen. That's basically it. No pbx user, then no way to go to voicemail, no way to schedule the ivr message for after hours, now way to handle a call when the user is busy, no way to do all the rest of the features.

You can use the ARS for ip address authentication by simply setting up an ARS with the top part all blank. Just fill in the in and out patterns. I love this feature and use it with my main carrier.

Concentrate on mastering the ARS, it can do everything you need and more. The premise is fairly simple, matching patterns of the caller and the callee and then performing the call on the right side deploy section.

To answer you question about the call directing there are a couple ARS solutions.

Matching

From: 0201231234@


Deploy

To: 101
Can be either a pbx user 101 which in the user area is forwared to 90201231232

To:90201231232
Can be directed directly to the destination number.

Notice the prefix 9. Assuming you have an ars set up where the user dials a 9 before the number. You must do it here also.

Actually in the past I always sent the call to a user where i there forwarded it to a number. It was only testing this for you that I realized i didn't have to do that, I can just put the number in the deploy directly. So, thanks for making me realize that.

Hey, i hope this helps and please don't hesitate to get more help here.

Nick

[colin2710]

Hi Nick,

Many many thanks for your post.

I think I can see the light!!!

I am just off for a long weekend break and I will post a full reply to your post on my return to the UK on Tuesday.

Many thanks again.

Warm regards - Colin

[colin2710]

Hi Nick,

I am sorry not coming back to you sooner, but my long weekend turned into a whole week!!! I am now back in the office and have been looking at your post in more detail.

We have purchased the PBX with the SIP server version 2.1.6.2 Basic

I think I understand a little better how the Brekeke PBX integrates with the SIP server. I thought that ARSs were for PBX users only - thanks for your information on that as it has helped a lot. In our case we would have very few PBX users but many SIP users, but we may need to upgrade to 40 or 60 PBX users later for another project we have later this year. Thank you for pointing out the number of concurrent users position with ARSs and I think that is logical.

As mentioned above, most of our traffic will Not need a PBX user or SIP User Authentication and most of what we want to achieve in phase 1 of our project is quite simple, for example one of our suppliers is the well known company Voxbone and all their calls come from a set range of IP address 81.201.82.21 to 81.201.82.33 (13 IP addresses), there is No authentication needed, they just send us the call as "442081231234@sip.oursipserver.com" (442081231234 = e164 telephone number), we then want to forward the call to a specific number via our VOIP termination providers as "442081231234@sip.theresipserver.com". These would need a dialplan for each inbound number, as there be about 200 of them. We have setup the SIP server to link to a SQL database and it work very well.

I have been able to get this to work with some of our calls but Not all, but I need to work on that one because some of our suppliers (I think) put different information in the headers.

We would also like to accept calls from certain IP addresses Only, in Voxbone's case, those 13 IPs and no others. I would be grateful if you know a way to only accept calls from the 13 Voxbone addresses and block all others in a dial plan or ARS.

I have done some experiments today with ARSs and I have solved 2 of my problems already, thanks to you. I will be doing some more experiments over the next few days and will report back to you, if I may.

Again many thanks for your help.

Regards - Colin

[voipwell.com]

Hi,

I saw on another post you were trying to forward a call to a voip provider that requires login authentication. The responder was correct that this can only be accomplished using the pbx ARS not the sip dial plan.

As to your other question about only receiving calls from voxbone address ending in 100-110 you need to set auth off on their calls so it would go something like this. This is assuming in sip config you turned auth on for invites.

Dial Plan

Matching Patterns

$request = ^Invite $addr = 111\.222\.333\.[100-110]

Deploy

$auth = false
To = %1@provideripaddress