Cisco CallManager to Asterisk Migration

Hi Everyone,

I am managing 300 user Cisco CallManager servers. After my initial test installation with Asterisk@Home, i am impressed by the feature set and planning to migrate to Asterisk@Home for my company. I got very less budget for this evaluation. I need some help from you experts to guide me. Below are my questions

  1. I see Asterisk@Home is replaced by Trixbox 1.0. Is this true? Do i need to go with Trixbox as Asterisk@Home development stopped?

  2. Can Asterisk@Home manage my 300 users load (I will choose good servers for this) Do i need multiple AH Servers? How do i integrate them?

  3. How do I configure DID Lines last 4 digits as my Extensions? (Incoming calls should come directly to user extension) This is very critical for me.

  4. I am planning to use 1 (PRI) Trunk for Incoming Calls and 1 for Outgoing Calls (PRI). I am planning to buy TE210P. Please suggest. (Does this card support PRI or i need order T1 Lines? I need more channels).

  5. Will i be able to integrate Asterisk PBX & Cisco at later time if reqired?

  6. How do allow international calling only to authorized users?

Please spare few minutes of your valuable time & help me in my project. I will prepare complete documentaion and share with Asterisk team members.

Thanks
Sekhar

Sekar -

I can answer some of your questions, but some of them will require a lot more reading on your part as well:

  1. As I understand it, Trixbox is the replacement for Asterisk @ Home so you would use Trixbox. All development work is going into Trixbox from what I have read.

  2. This depends on what exactly you will be doing on the * box. Are you going to use a SER or try and terminate 300 sip devices directly on your * box…? As far as integrating multiple * boxes, there are different ways of doing so depending on your desired outcome. If you are attempting multi-site SRST that is one thing vs load balanced local boxes.

  3. This is done in the dialplan and it is very simple. One way is that you create an inbound entry for the DID and have it use a Goto into a menu for another extension, or just have it dial the extension directly. Again, this is something that you can do different ways, but it is easy to do.

  4. Digium has PRI cards that will support upto 4 PRIs. Take a look at digium.com

  5. CCM has the ability to utilize SIP as does * so I would imagine the “general” answer to this question is yes, but it will depend on what type of integration you want to attempt with your CCM system. The other part is how well CCM actually utilizes SIP for its everyday operation.

  6. One way this could be done is to use different contexts within your dialplan (extensions.conf).

What you are looking to do is not a trivial task. I would suggest spending a lot of time reading and setting a test box or two to try and figure out if * will do everything you need it to do in regards to integration with your CCM.

Hope this helps.

Hi Richard,

Thanks for your valuable inputs. I will install Trixbox1.0 today. I am planning to do load balancing at this time. Multiple sites are not my goal for now.

Please suggest me some documents/links to understand more on Direct dialing configuration (DID) & International restrictions. I am reading AH manual (2.8 version).

My goal is by this weekend, i want have VOIP Server ready for SIP Physical phones (5) & 5 Soft phones. My Cisco phones are capable of 3 lines (7940), how do i activate line2 thru Asterisk? I will keep my update to this thread. Please input me your suggestions.

Really appreciated your time.

Thanks
Sekhar.

[quote=“sekhar”]Hi Richard,

Thanks for your valuable inputs. I will install Trixbox1.0 today. I am planning to do load balancing at this time. Multiple sites are not my goal for now.

Please suggest me some documents/links to understand more on Direct dialing configuration (DID) & International restrictions. I am reading AH manual (2.8 version).

My goal is by this weekend, i want have VOIP Server ready for SIP Physical phones (5) & 5 Soft phones. My Cisco phones are capable of 3 lines (7940), how do i activate line2 thru Asterisk? I will keep my update to this thread. Please input me your suggestions.

Really appreciated your time.

Thanks
Sekhar.[/quote]

Not real sure on the 7940, but on my 7960’s each line can be configured for a different extension or even a different * box.

As far as what to read on configuring your DIDs, you should become very familiar with the voip wiki. This is a quick link on DIDs:

voip-info.org/wiki-Asterisk+tips+DID

as far as creating call security, etc - start here:

voip-info.org/wiki/index.php … sions.conf

Cheers

Hi Richard,

Can you please send me a sample file for 2 line cisco7960 configuration. Can other lines be used when line1 & line2 etc. are busy?

Also, If i take VOIP account from broadtalk.com, will i be able to make multiple calls?

Please send me sample exten. conf file for DID Lines. Will users be able to call each other by 4 digit number?

Thanks
Sekhar.

Hi Richard!

sekhar,

The 7940’s are the same as 7960’s.

Basically to configure a second “line” you simply create another extension in your asterisk configuration with its own username/pass etc. You can mentally treat this for simplicity as if you have 3 seperate phones…
Asterisk would be configured for three “accounts” and the phone would register under all three of those accounts.

The reason the phones work like this is because you can have not only 3 lines, but 3 seperate carriers if you prefer.

Are you in need of any aggregation? Call origination or call termination?
I can get you going on DID’s and etc without the need for PRI’s and all
that. Were selling SIP trunked DID and other services. We could also
help you learn Asterisk on the quick path. One of our engineers could
assist you and could save you time?

Let me know…

chris_jester@suavemente.net

Cheers!

sekhar,

TrixBox uses FreePBX 2.1 to create the dial plans. In the FreePBX extension setup there is a Direct DID field. All you need to do is enter the DID in that field and you’re set.

For International call in FreePBX you can setup an outdound trunk that matches the numbers dialed then set up a PIN Set (and even tell it to enter the PIN used in the CDR) for use with the outbound route and give the users a PIN to use when they place an international call. If they have a PIN the call goes through if not it doesn’t.

All of this can be set up very easily using TrixBox/FreePBX.

Tom

OK Asterisk is rock solid no two ways about it.

However TrixBox is still BETA, the folks building did a complete turn around in the way things are done.

before you jump into a beta software where you are always looking for the next fix do a little reading. (go read the trixbox forum)

Now I am not a Asterisk / linux Guru so I like my stuff to work.

I can tell the asterisk on centos 4.3 with FREEPBX 2.1.1 works great, the folks who are building it or great.

I just see a few month for trixbox to get out a stable bug free (Ya like we see that with any software)
A big got-ca on Trixbox has been zaptel.

I had little problem when update linux to 2.6 but nothing like the trixbox users had… I have centos 4.3 and asterisk zaptel from source and it was easy for me to “fix it” as the trixbox is doing so funny stuff with yum down rpm’s or something, folks are having a hard time.

Thanks to everyone,

For last couple of weeks i struggled to install zaptel and finally yesterday i was able
to do so. I still see “UNCONFIGURED” when i run zttool command. i guess i am missing some ore configuration.
I am not using Trixbox, so can some help me in getting the configuration done thru manually for
incoming DID lines? any sample configuration files will be helpful.

i got te205p card, i want to use trunk1 for incoming and trunk2 for outgoing.

tletourneau: Thanks for the advice, i am afraid of the BETA version of trixbox, would request you
to send me some sample conf. files if you have for manual configuration.

Thanks
sekhar.