Sunday, July 30, 2006

Connecting two Linksys ATAs - Point to Point without SIP registration

This will create a line extension between a Sipura 2100 (although 2000, 2002 & 1001 would be the same) & a SPA3000 connected either to a plain-old telephone line (POTS) or an analogue PBX system. From the looks of the Visio- This is what you are trying to accomplish – There will be no registration to a SIP Proxy, addressing will be handled by the IP addresses.



Incoming calls on the SPA3000 PSTN/PBX line will ring the phone connected to the SPA2100.






IP address of the first SPA3000 is: 192.168.192.10

IP address of the SPA2100 is: 192.168.192.12



These addresses are examples. Replace them for whichever addresses you've configured your ATAs to. It would be a good idea to configure them with static addresses rather than using DHCP since we'll be using the IP addresses to contact each ATA. If these addresses were to change, you would have to reconfigure them- But you knew that.



How to:



SPA3000: Login to the web interface by typing its IP address into a PC connected to the same subnet & click “admin login” & “advanced”.



Go into the tab for the “PSTN Line” and change the settings below:

“Make Call without Reg” to “yes”.
“Ans Call Without Reg” to “yes”.
Change dialplan 2 to read “(S0<:192.168.192.12>)”. This sets up a hotline which calls the remote ATA using its IP address at the standard SIP port of 5060. It's wise to leave use dialplan 2 (or any number after that) & not dialplan 1 since most of the default settings on this page are set to use dialplan 1.
“PSTN Ring Thru Line 1” if you have a phone connected to the 3000 & want it to ring when a call is received via the PSTN, leave this as “yes”. If you only want the phone on connected to the remote ATA to ring, set this to “no”.
“PSTN Caller Default DP “to “2”. This should match whichever dialplan you setup two steps ago. Again, try to avoid changing dialplan 1 in this case.
“PSTN Answer Delay “change this to zero. Otherwise any incoming calls will not be forwarded to the remote ATA for 16 seconds (as the default value here is 16). However this will cause the phone connected to the SPA3000 to not ring even if you specified “Ring Thru Line 1” above. Adjust the delay accordingly if you want the phone to ring before the caller is forwarded onto the SPA2100.


At the web interface - In the “Line 1” tab, change the settings as follows:

“Make Call without Reg” to “yes”.
“Ans Call Without Reg” to “yes”.


Now at the web interface - in the “User 1” tab alter the following fields:

“Cfwd All Dest” to “gw0”. This forwards any incoming calls on the VoIP line to gateway- 0 which is the FXO connected to the PSTN. Now when the SPA2100 calls the IP address of this 3000, it is automatically forwarded to the PSTN & the user of the remote hears the PSTN dialtone.


Now you are ready to set up the 2100…



At the web interface browse to the tab for “line 1” and alter the following settings:

“Make Call Without Reg” to “yes”. This allows calls to be made without being registered to a SIP registrar.
“Ans Call Without Reg” to “yes”. This allows calls to be received without being registered.
“Enable IP Dialing” to “yes”. This enables the ATA to dial using IP addresses rather than SIP URIs which is ideally what should be used.
Change the dial plan to read “(S0<:192.168.192.10>)”. This sets up what is called a “hotline”. So when the phone is picked up, it automatically dials the number (or IP address in this case) without the user doing anything. In the .e4 example it connects to the first SPA3000.


Now we have to secure this setup… If you are worried about Mike, myself or anyone else abusing this configuration, we could input the IP address of the

SPA3000 into an IP phone & make calls to china or anywhere else.



So……

1 Comments:

Blogger Marco Domingues said...

How can i configure to acess the the second line of sipura 2100 ?

5:11 PM  

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