Can You Configure Cisco Phone Spa525g2 to Work on 3cx?

SPA3102 with Freepbx setup (Singapore)

Purpose of the certificate

I was tasked to implement VOIP system in a pocket-sized company with almost 20 staff in Singapore and about 10 staff in India role.

To empathize how VOIP work, and how VOIP can communicate with Analog phone arrangement, I have purchase the Linksys SPA3102 ATA (About Singapore South$90 From Sim Lim Mediapro), try to understand how everything can work together.

I have tried to google for document how SPA3102 work in Singapore environment, but without success. I managed to find data on how to setup SPA3102 with Freepbx, simply document was long and not very piece of cake to read and follow. So I decided to write this document, hoping anyone like me, existence chore to setup VOIP, tin follow and empathize.

Purpose of the Lab setup

The purpose for this lab setup, is to install FreePBX, with few extension number, and I have a dwelling analog line (PSTN line), and wanted any of the few extension number from softphone able to make a call out thru this analog line. Objective two is off course to permit incoming call from analog line, to go to an Interactive voice respond carte, and select the pick, and forward the call to the selected extension or softphone.

Rest of the FreePBX feature, is non in this lab telescopic, and you should be able to find a lot of data on asterisk feature.

There are also a lot of document covering on SPA3102 to connect to SIP service provider (In Singapore, service provider like hoiio), this certificate will not cover those too.

Setup step (Summary)

ane.

Setup the hardware (SPA3102)

b.

Setup Outbound Route

c.

Setup Inbound Route

6.

Configure SPA3102 to link to FreePBX

Hardware Setup

I presume that you lot have some network setup noesis, setup the hardware equally below.

1.

Once above is setup, power on the SPA3102 device

2.

Using the counterpart telephone, press "****", and after the greeting bulletin printing 110# to remember the Net/WAN IP accost. Take a note of this IP  Address.

3.

Printing "7932#" and so press "1#" to enable the HTTP Configuration over the WAN port. At present, press "one″ to save.

4.

Now point your browser to the gateway device's configuration folio: "http://x.x.x.x", where "x.ten.ten.10" is the IP Address assigned to the device past the DHCP Server.

5.

One time the page loads click on the "Admin Login" link on the acme right mitt side of the page. This will provide the basic administration configuration version of the interface.

6.

Now click on the "advanced" link on the top right paw side of the page. This will provide the advanced administration configuration version of the interface.

7.

If you are able to admission to the above, put the hardware i side, and setup the FreePBX.

FreePBX setup

1.

Create a virtual machine for the environment.

three.

Follow the step by footstep installation

five.

If yous are able to access to the folio, setup the extension number.

Extension Number setup in FreePBX

1.

Become to Application, click on Extensions

two.

Select "Generic SIP Device" and click "Submit"

3.

Select "Generic SIP Device" and click "Submit"

4.

Full in only the important fields

c.

Secret  (Password)

6.

Click on "Utilise Config" on the top.

7.

Repeat above to create a few more extension number.

8.

Next gear up to utilize the extension using the softphone.

Install Softphone (1 on PC, and one on Smartphone)

ane.

On the windows Bone, install x-lite

2.

Configure the xlite account

iii.

Key in the Account Name (what key in FreePBX)

iv.

Key in the User ID (Extension Number in FreePBX)

5.

Central in the IP address of the FreeBPX server

vi.

Key in the Password (Secret in FreePBX)

8.

Install some other xLite on another PC, or install Zoiper on Smartphone (Android)

9.

If y'all are setting up Zoiper for another extension, the setting are similar, Account Name, Host, Username (Extension in FreePBX) and countersign (Surreptitious in FreePBX).

10.

Test to call from one extension to another extension, and see the call go thru.

11.

If the call go thru, y'all are prepare for the adjacent step.

Configure FreePBX – SIP Trunk (for SPA3102 Analog Line Connectedness)

Extract from http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx

ane.

Create a new SIP trunk in FreePBX.

ii.

Go to Connectivity and click on Trunks

three.

For Caller ID put my inbound DID From Singtel POTS line.

4.

Max channels, put 1 since information technology's only i line. This is important because if y'all don't do this, and this torso is decorated, calls may not fall through properly to the next trunk.

5.

Trunk proper noun: Call this 1-pstn

vi.

Outgoing settings - Peer settings as below

disallow= all

allow= ulaw

canreinvite =no

context=from-trunk

dtmfmode =rfc2833

host=dynamic

incominglimit =1

nat =never

port=5061

qualify=yes

secret=password

type=friend

username=1-pstn

Few important setting in a higher place will need to be the same in SPA3102 PSNT line to connect to FreePBX setting

When configure the above in SPA3102, will demand to make sure the setting is the same.

vii.

Click on "Submit Changes"

8.

Click on "Apply Config" on summit

Configure FreePBX – Outgoing Route (for all extension to call out thru analog line)

Extract from http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx

i.

Create a new Outbound Routes

ii.

Name of the route : Spa3000

3.

No password for the route

4.

The important setting here is the Dial Patterns. I have only configured to dial 9, to call an external number. Example, on the softphone dial 991231234, will put a call to PSTN line "91231234". If you dial normal extension number, information technology volition nevertheless go to extension number.

5.

Prepare the Trunk Sequence 0 to use "1-pstn"

half-dozen.

Click on the "Submit"

7.

Click on the "Apply Modify" on top.

Configure FreePBX – Incoming Route (for incoming call from analog line to go to one fixed extension)

Extract from http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx

1.

Create a new inbound Route

2.

Name of the route setup as "1-pstn"

3.

DID number gear up to the DID number, this number must EXACTLY match the number you put in "Punch Programme 2" in SPA3102 PSTN Line setting

4.

Set Destination as extension number to go to, or setup every bit IVR.

5.

Submit Changes and call back to click the "Employ Change" for the update.

Configure SPA3102 to link to FreePBX

i.

Browse to the SPA2102 admin page, and get to advance setting

SIP Tab

ii.

Before you exercise anything else, go to the SIP tab. Wait under RTP Parameters and bank check the RTP Package Size. Linksys has set up this to 0.030 by default, which is not correct for utilise on ulaw (G711u codec) connections. Change it to 0.020 instead (or 0.02 on older Sipura devices). If you don't do this, you may experience strange problems with "choppiness" or random clicks on some calls only non others, and you may also feel problems when playing Asterisk sound files.

3.

Adjacent go to PSTN Tab

PSTN Tab

Proxy and Registration

one.

Line enable set to Yes.

2.

SIP Port is set to port 5061 (same as FreePBX port number in the trunk setup)

three.

Proxy fix to the FreePBX IP address of your Asterisk box

4.

Make Call Without Reg: Yes

5.

Ans Call Without Reg: Yes

6.

Register Expires: 300

Subscriber Information

1.

Display name: Put something here that will identify this line - this is simply displayed on your phones if you get a phone call with no Caller ID data (or you don't subscribe to Caller ID). Keep it at 15 characters or shorter. You could apply something like LOCAL PSTN Phone call.

2.

User ID: ane-pstn ; very important - this must exactly match the FreePBX Body proper name and username in the trunk configuration!

iii.

Password: XXXXXX (aforementioned as you used in FreePBX Trunk settings).

Sound Configuration

1.

DTMF Process INFO: Yep

2.

DTMF Process AVT: Aye

3.

DTMF Tx Method: Motorcar

Dial Plans

1.

Under Dial Plans it'southward important not to change the default (twenty.) on whatsoever except Dial Plan 2.

2.

Dial Programme 2 set every bit  (S0<:1234567890>)

3.

Supervene upon 1234567890 with the telephone number of the PSTN line coming into the device. Notation that this must exactly friction match the DID number in your FreePBX Inbound Route setting for this device. If the number here and in the Entering Route don't friction match exactly, you lot won't receive incoming calls.

VoIP-To-PSTN Gateway Setup

This is some other of import settings section.

1.

VoIP-To-PSTN Gateway Enable: yeah

ii.

VoIP Caller Auth Method: None ; use "None" to start, you tin can change afterward for added security

3.

VoIP Pin Max Retry: 3

4.

One Phase Dialing: Yeah ; very of import

5.

Line 1 VoIP Caller DP: none

6.

VoIP Caller Default DP: none

seven.

Line 1 Fallback DP: none

PSTN-To-VoIP Gateway Setup

This is another important settings department.

1.

PSTN-To-VoIP Gateway Enable: Yes

2.

PSTN Caller Auth Method: none

three.

PSTN Ring Thru Line i: no ; I use Asterisk for my routing.

4.

PSTN Pin Max Retry: three

five.

PSTN CID for VoIP CID: Yes if you subscribe to CallerID service on your PSTN line, otherwise No

half-dozen.

PSTN CID Number Prefix: (Leave Bare)

7.

PSTN Caller Default DP: ii ; important - here is where information technology sends the calls to.

8.

Off Hook While Calling VoIP: No

9.

Line 1 Signal Hook Flash To PSTN: Disabled

ten.

PSTN CID Proper name Prefix: (Exit Bare)

xi.

Exit everything else in this section blank.

FXO Timer Values (sec)

1.

Voip Answer Delay: 0 (The original recommendation was 1, only this can cause a spurious half band on outgoing calls, before bodily ringing from the called line commences, so 0 is now the recommended value).

2.

PSTN Answer Filibuster: If y'all do not subscribe to CallerID service on your PSTN line, this tin be fix to 0. Most users will want to set it to at to the lowest degree 3 so that the incoming CallerID data is captured. In rare situations you may need a slightly longer delay (5 should be more than enough).

PSTN Disconnect Detection

1.

For Singapore Singtel Phone line, please modify Discount Tone to

425@-30,425@-thirty; two(0.75/0.75/1+2)

International Control

Bank check the settings hither - each country uses different values for PSTN lines. If yous alive in Australia, Canada, the United States or well-nigh other countries with modern telephone systems you probably won't take to change anything except peradventure the gain levels, so we'll only deal with them for now. The default values for both the SPA To PSTN Proceeds and the PSTN To SPA Gain are 0 (zero), and that's where you should leave them when y'all're showtime setting upwardly the SPA-3000/3102. Only just so you know, hither's some information on those settings:

1.

If the SPA to PSTN gain is set too low, the parties on the PSTN side of the connection will probably complain about your book being likewise low, or will inquire you to speak upward or talk closer to the telephone. If it is set too high, however, you are more likely to hear echo, and outgoing calls may fail because the level of DTMF tones sent by the SPA-3000/3102 volition be too "hot" to annals properly at the PSTN switching equipment.

two.

If the PSTN To SPA Gain is set too low, you lot'll hear low volume levels on PSTN calls. If it's set likewise loftier, the people on the PSTN side of the connexion will be more likely to hear echo (they may hear their ain voices echoed dorsum from your stop). Also, any echo that has been reflected back to you will be heard at a higher volume level, and volition therefore exist more objectionable.

iii.

While the default levels are commonly acceptable, we plant that boosting both values up to 3 produced a more "natural" sounding book level in both directions. Yet, this is very much dependent on the characteristics of the PSTN line - if you're on a very short loop, values of 0 may be acceptable for both settings, if on a very long loop you may need to go even college than 3. The valid range is -15 dB to 12 dB in one dB increments (just just enter a numeric value, practise not enter "dB" in the text field). If you lot take actual test equipment available you can fine-tune the book settings for best results.

four.

I have non play with this number withal, but gain of zero is ok now.

Exam incoming and outgoing line

1.

Call you DID analog number, and check if it go to the extension or IVR.

ii.

From your softphone, call 9xxxxxxxx, and run into if it call the external number.

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