I was hired in a new company that hasn’t had an IT team for years. Things are working in this company as ‘magic’ without maintenance. Since VoIP is under my authority, I was inspecting the telephony system we have in place, and am interested in moving out to Asterisk.
The Cisco Call Manager I m using (3.0) is setup in a weird way.
As you can see, We’re just using the CCME to access the PSTN. No practical VoIP is in place (we dont have remote offices).
The problem I m facing is that, in all my current Asterisk Experience, I always install some type of FXO card on the box to interact with POTS. in here, there’s a router and a modem, and I m not sure how I should proceed. It just seems to be illogical, I dont know.
Any help will be highly appreciated.
Thanks a lot.
Thanks for the prompt reply. Nope, no VoIP provider at all:
-All local communications are routed through the local phone company
-All the international calls are routed through … the local phone company
The CallManager is simply useless. its acting like an old school PBX, with a nice database that stores all phone call logs. nothing more to add.
Nobody has a clue why CCME is there, and everyone wants all the nice features asterisk has to provide!
Yet, I dont understand the usual setup this should be in. I was expecting CCME to look like asterisk from a design POV. The only viable configuration I found in the Callmanager is that all ‘route plans’ are set to use a ‘gateway’ - which is the internal ip address of the router.
Can I have something like that in asterisk ?
A trunk that sends all outgoing calls to a specific IP address?
If that’s the case, then i believe I can substitute the call manager pretty easily
I will be sniffing the traffic moving between the callmanager’s interface and the router’s interface to see which protocol is in use. maybe this will shed some light on the topic.
Thanks for the reply once again! Everybody’s help is highly appreciated…
This sounds like a normal installation. The callmanager will be using the I/F card installed in the router for its out/in pstn traffic and the Call Manager is doing the rest. Call plans phone features etc.
Well except that ALL the phones are IP phones and the traffic to the router is IP.
The SHDSL “modem” will more than likely be a T1/E1.
Leave well alone and get someone in who does understand what is there. as It all seems very normal and can give the company an honest rundown on what they have and what is needed to be done.
Thanks for the reply. Yet, I have to personally take action.
We’re having a new branch office opening soon, and I cant have a Cisco Call Manager in the remote location due to fiscal issues.
I also cant have the CCM upgraded to support SIP. If this was the case, I would have let the the CCME in my HQ, and had an asterisk box in my remote location, and just created a SIP trunk. Alas, I can’t.
Managers know that they already have ‘this VoIP thing’ they bought a couple of years ago. They dont wanna invest in a new Cisco box, since 'VoIP allows them to talk for free". See the mess I’m in.
Yes, the traffic internally is all SSCP and IP, but as long as its in a LAN, i dont see the practical side of it.
And finally, There are no asterisk designers/consultants in my region.
I’ll appreciate any lead that anyone can post.
The modem has a CAT cable(not a WIC) running out of it, going to the router.
Can I take this cable and simply insert it in a second NIC installed on an asterisk box?
This way, I’ll have
it is definately an ISDN (BRI/PRI) device. I don’t know what the ‘modem’ is, you should find that out. It may be some kind of Virtual T1 device.
You should start by looking thru the 3600’s configs and figure out what it is set for. You may also want to call the provider (figure out who that is too) and ask them for more info. Your best bet would be if you can get the provider to send you a full detailed list of what all is being provided and where.
What I can tell you with relative certainty- the cable going between modem and router is carrying an ISDN signal of some kind, either BRI or PRI. You can interface Asterisk to a PRI with any one of a number of cards. From the modem to wherever it gets service from could be anything, but I will assume it’s also a PRI. If it is, you may be able to drop it entirely.
You will want to plug the PRI straight into Asterisk, and configure it in zaptel. Asterisk can use both the data and voice side of the PRI and route the data end over Ethernet. You do not want to plug the PRI into an ethernet jack.
Yes, I have experience in VoIP, I do realize that VoIp is the encapsulation of Voice data in IP packets to be passed on a packetswitchin network.
The text between quotation mark would be my managers knowledge - they have no clue what it is, and they want it to happen.
Alas, CCME v3 has no support for SIP, no.
My network design is simple, I dont want to loose you
HQ ---- internet ---- a new remote office to open in the UK.
They want to make calls from the HQ to the remote office. they dont wanna pay for international charges (that’s where the ‘free’ part they have in mind comes from).
HQ has CCME3, and the remote office can be running anything. exept no budget exists.
Integrating CCME3 with asterisk will help me. I wouldnt have to sbstitute the cisco infra i have in HQ, just connect it to the remote office.
I found a tutorial on voip-info.org regarding this. It mentions the requirements are at least CCME v4 to have SIP enabled.
Alas, for monetary reasons once again, I cant upgrade my box from v3 to v4.
Any help is appreciated.
Hi I think you need to work out the cost of the V3 to V4 (or v5) upgrade verse the cost of a new E1 card, server, Sip firmware for the sets and the time to reflash them and then the time and cost of changing over to an * with the same feature set. I think you may find that the Upgrade may come in less in both time and trouble. Then you just need to deploy an * in the UK office and connect up via SIP
Which ever way you go it will cost money. THen the final thing is to work out what the capital cost verse using a cheap calls supplier would be.
You could just add a e&m card to the 3600 and interface that to an ATA (Multitech) support e&m then route that as SIP to the UK *
There seems to be nothing unusual in the config, other than a lack of understanding how to go in and look at the details to understand it and extend it. Here are a few questions to gauge whether or not you should bring in some extra help:
Is it Cisco CallManager Express running on a module of the 3600 series (I’m not sure if CCME works on older 3600 chasses, but it is an option on the newer 38xx series), or is it a separate Windows server?
Do you know how to log into the router and see if the router is using MGCP or H.323 to talk to CallManager if it’s an external CallManager?
Do you know how to login to the CCMAdmin pages in CallManager web GUI to look at the configuration of trunks, endpoints, gateways, and configure calling search spaces & partitions, etc?
Do you know what TDM modules are installed in the 3600 series? Can you see dial peer configurations to know what digits are pointing where?
Do you know how to tell if a call is failing because of an IP packet/routing issue or a TDM signaling issue?
Based on the lack of very relevant details related to the questions you are asking, and the lack of articulating the proper questions to ask to get this figured out, my bet is you need to find someone who can give you hands-on help if you have a short term deliverable. If this is just not an options, then you have a lot of reading to do. Nothing wrong with that, just don’t underestimate how much there is to learn, but make sure to enjoy the process!
Go to Cisco’s website, they have tons of docs that are more useful than asking random people on the Internet (but you have to start somewhere!). Here are a few links:
Overall link to voice products/tech support: cisco.com/en/US/products/sw/ … _home.html
None of these are blind “point and click, follow the directions” manuals. You have to understand the underlying protocols, the framework for how the devices interconnect, and these manuals each show bits and pieces of the grand plan that you have to piece together over time with experience.
Your replies are appreciated to an unimaginable extent!
Yes, it is running on a windows 2000 server.
No, for the simple reason that the router’s password is lost (i’m contacting a cisco partner to have this hopefully solved)
I convinced management to remove the Callmanager out of the network since it already depreciated, and go for the ‘smaller’ cost asterisk, and they are already all into it. I can’t find anyone that can give me hands-on help in this scenario, but i am willing to learn, enjoy, and hopefully at some stage have this solved!!
I will post any new update as soon as I have it.
I can’t express my gratitude enough. Thanks a lot guys !
I know i’m rushing things up, since my last post was saying that I’ll spend time reading about CCM and understand what is going on, but while browsing the net, I came over a product redfone produces, the foneBRIDGE http://www.red-fone.com/fonebridge.html
The produce seems interesting, it terminates E1 lines into ethernet.
Naive Question, but is this what my Cisco 3600 router is doing?
If so - from a highl evel point of view - can I replace the router with a foneBRIDGE, and replace my CallManager with an asterisk box?
I have setup a 3 callmanager (version 3) and one Asterisk box integration. Although SIP is not supported on the callmanager v3, h.323 is. H.323 is also supported on Asterisk with its add-on. I have three gateways: one MGCP, and two h.323. The Asterisk server connected to each of the callmanagers through an h.323 connection, and connected directly to the PSTN through the two h.323 2600 routers.
You can tell if your router is being used as a PSTN gateway by checking inside the callmanager administration page. It will be under the devices>gateways section. The icon to the left of the gateway will indicate what type of gateway is connected. If it is using h.323, then you can just setup your Asterisk box to the router. If it is MGCP, you can change the config so that the callmanager is connected to the router using h.323, instead of MGCP.
I have integrated both callmanager and Asterisk. I would caution in being so quick to get the callmanager out of the picture. I would strongly suggest setting up a test server with phones connected that can place calls to and from the current callmanager. If this works, it would solve your issue with the remote location. If your company is happy with the performance of the remote location, then you can duplicate the setup in your main location. In my opinion, it takes vastly more experience to get Asterisk working in a live business environment.
Purchasing support for either the Callmanager or the Asterisk box could be the best suggestion you make to your bosses. The risk of not doing so solely rests on your shoulders. That is more risk than I am personally willing to take. At least propose it.