Newbie, how to integrate to existing PBX system

Dear All,

I am very new with this, I have installed AsteriskNow, how to integrate to existing PBX system? …

Please advise.

Many thanks in advance

Regards

Winanjaya

Well you can integrate it, or you can replace it completely w/ asterisk.

If you really wanna integrate it, get a few ATA devices and hand them extensions through asterisk, I did that a while back when there was no funding for phones.

Now that were using SIP phones, the old pbx goes in the trash.

Thanks, could you please advise what brand/model of devices should I use?

Thanks
Winanjaya

BTW, can I use linksys FXO to integrate to the existing PBX?

Any ATA device with an “FXS” port will do.

If you have POTS lines coming into your existing phone system (that are not PRI) you can simply use the TDM400P card with a few FXO modules.

Lets say you have 4 POTS lines coming in, you can get a TDM400P with 4 FXO modules, plug the analog lines in there and use that for your incoming.

Then you can take your FXS ATA devices and configure them with extensions to asterisk.

Then take the lines from the FXS portion of your ATA device into your existing phone system.

Create a dialplan that rings out to your ATA extensions, thus sending the FXS signal to your existing phone system, thus ringing at your communications terminal.

You can do the same thing with Voip lines, pri, etc.


If you plan on using just asterisk you can get a hold of some pretty good grandstream SIP phones that will do the job, Here at the office we use GXP2000 and they work great. It took some time and some tweaks but we are very satisfied with the result considering we only spent about 1500 bucks for an entire system (including the Server, the TDM card w/ modules, and the phones) We have a total of 15 phones, 2 POTS lines, and 4 Voip lines.

Something to think about, there is no limitations in asterisk, there is no one telling you that if you want a new feature it will cost 5000 dollars either.

There are also hardware drivers for meridian phones among others

Let me know

Thanks, now I am trying Linksys SPA-3102 to connect to PSTN, could you please advise … why I got “Service Unavailable” on my X-Lite softphone, below are what I have done.

AsteriskNow:

Service Provider created:
Provider type = Custom VoiIP
Protocol = SIP
Registered = Checked
Host = 172.16.1.169 (The IP address of the Linksys)
User Name= 250 (my extension no)
User Password = my password

Calling rules:
DialPlan2 created
Place this call through = Custom - Linksys
Dialing rules = if the number begins with 0 and followed by 9 digits or more
Strip 0 … blah blah

And at the end, I assigned user 250 to use Dial Plan = DialPlan2

What I missed?

Please help

Thanks a lot in advance

Regards
Winanjaya

explain to me what exactly you are trying to do…
you have an existing phone system…
you have Linksys SPA-3102…
you have a standard line line…

do you have a voip provider?
does your xlite phone communicate with the asterisk server?

You wrote:
Explain to me what exactly you are trying to do…

==> Ok, I want to make outgoing call (ie. I call my cellular phone at 0817123456)

You wrote:
you have an existing phone system…, you have Linksys SPA-3102, you have standard line…

==>Yes, but currently I just want to make outgoing call via Linksys and my Linksys SPA-3102 already connected to direct PSTN line and it also has a single line phone.

You wrote:
do you have a voip provider?
==> No

You wrote:
does your xlite phone communicate with the asterisk server?
==> Yes, I tried to installed xlite on the other PC and they can communicate each other.

What I have done are as follow:

I created ext. 998 password 1234
I created ext. 999 password 1234

refer to Line 1 tab in Linksys SPA configuration:
Proxy registration ==> 172.16.1.74 (this is the IP address of AsteriskNow)

Subscriber information:

user id ==> 998
password ==> 1234

refer to PSTN Line in Linksys SPA configuration

Proxy registration ==> 172.16.1.74 (this is the IP address of AsteriskNow)

Subscriber information:

user id ==> 999
password ==> 1234

I tried to call 998 or 999 from X-Lite … it displayed “Connection Established” but the phone does not ringing

I also tried to call my cellular phone (0817145779) from X-Lite … but it displayed “Service Unavailable” …

as newbie, I thought I missed something in my AsteriskNow…

please help me out from this problem… I have stuck on this for almost 5 days, I asked to Linksys but they answer this is the registration problem of the AsteriskNow…

Many thanks in advance

Regards
Winanjaya

FYI,

sip.conf

context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
allow=h261
allow=263
allow=263p
allow=h264
videosupport=yes
register=999:1234@172.16.1.169/999
register=998:1234@172.16.1.169/998

Citel’s TVA can take Nortel, Siemens, Ericsson, Avaya, NEC, Panasonic, Toshiba handsets and convert them into SIP based IP phones. The TVA is a device that you can plug the amphenol connector from the PBX and you would disconnect the amphenol connector and plug it into the Citel TVA and those digital handsets would then be able to connect and register with Asterisk. Is is very similiar to an ATA but is the only one that can take Digital Phones and convert them into SIP devices.

citel.com/Products/Portico.asp