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[Asterisk] Trying to configure my Linksys PAP2 with Asterisk

I'm not that familiar with BroadVoice, but why wouldn't you want to set it up as a trunk on your Asterisk box? Then your PAP2T would only need to connect to the Asterisk server.

You can use inbound rules to make sure any incoming calls on the BroadVoice account only go to that one line.

You should be able to make this work. Set up the PAP2 with a static local IP address. Configure an extension on Asterisk that matches the Broadvoice user ID and password, with a hard-coded local IP address, so the PAP2 won't have to register there.

Incoming calls sent to that extension should ring the PAP2 with no problem. Then, set up the PAP2 dial plan to route some patterns to the static private IP address of Asterisk. You can probably work out a plan that won't require a timeout (or dialing #) for internal or domestic calls. For example, 01+extension (internal); 0+areacode+number (domestic); 00+countrycode+number (international). Assuming that you don't have extension numbers beginning with 1 (or if you do, have the 01 include that), then this should not conflict with the Broadvoice patterns.

The PAP2 registers with Broadvoice normally -- it won't be registering to Asterisk at all.

Incoming calls from Asterisk intended for the PAP2 will be sent there by having the PAP2's IP address hard-coded in the sip.conf file.

Outgoing calls to Asterisk will be sent by having some PAP2 Dial Plan entries hard-coded to the static private IP address of the Asterisk box. However, Asterisk will still try to authenticate the call (and the PAP2 has no other credentials), so Asterisk should be set up to accept the Broadvoice user ID and password.

It should be noted that in order for this to work, "Restrict Source IP" needs to be set to "no" on the PAP2. If you are forwarding port 5060 in at your PAP2 from the Internet, that means that anyone who stumbles across your pap2 in a scan can ring your phone.

See: »[Future9] Future-Nine oddities

reply to wifi4milez

In a typical PAP2 setup, SIP Port is set differently for the two lines. An incoming call will ring the line corresponding to the port to which it is sent.

Possibly, the Line 2 extension is confusing things; try unregistering it (just set Register to no in the PAP2). Then, set up a new extension with host pointing to the PAP2. If it doesn't work, see what SIP is being sent out (if none, there should be an obvious error in the console log) and what response, if any, is being returned by the PAP2.

reply to wifi4milez
I like to tinker with my PBX and want to make sure I don’t accidently impact my ATA, so I’ve tried different setups. I originally took the extension path making one with matching username/password, but I currently have some characters in my primary username and sadly, FreePBX doesn’t allow characters in an extension. So, I now use a trunk for calls from the ATA and a custom extension for calls to the ATA. I’m sure there’s many ways to accomplish what you want, but below is something that should work for you.

As an example, let’s assume the following:
Your ATA is set with an IP of 10.1.1.2 and your line is using port 5061 and the primary user is 12345_ATA. And your PBX is 10.1.1.3 and the bindport is 5080. You dial #9 then phone number (and then # to avoid any delays) to route from the ATA to the PBX. Calls to extension 4444 on your PBX go to the ATA.

On your ATA, modify your dialplan to include the following:
|<#9,:>[x*].<:@10.1.1.3:5080>|

On your PBX, to allow a connection from the ATA, create a SIP trunk with a name of your choice and include the following peer details (since my ATA and PBX is isolated, I authenticate based on IP and port):

host=10.1.1.2
port=5061
type=friend
callerid=PAP2 ATA <4444>
nat=yes ;might not be needed, depending on your network
insecure=port,invite
context=from-internal

»www.voip-info.org/wiki/view/Aste···sip.conf is a good wiki for Asterisk settings.

For calls to extension 4444, setup a custom device. In the device options section, use the following dial string:

SIP/12345_ATA@10.1.1.2:5061

reply to wifi4milez
The easiest and simplest solution for tinkering is to go buy a cheap phone for $10 at Walmart.
This is very foreign to me, as I have been using asterisk for the better part of a decade - to me the PAP2 is simply an extension device of the * server, and nothing more (not involved with any outside provider).
So, I had to think about it for a few minutes. Whatever you do will only be temporary. Once you get comfy with asterisk, you are going to put all your trunks on the * server and use the PAP2 as an ATA for the * only.

Does Broadvoice have inbound SIP uri? Many providers do b/c inbound SIP uri is of little cost to them compared to inbound calls from POTS.

What you do is set up an extension in asterisk that forwards to the sip uri to reach your Broadvoice account. KISS.

The drawback is that you will only have inward calls from your * server, but not outbound. It's basically like "call forwarding".

If you want to really KISS, you can just set up an extension to forward to your BV number. You will be using minutes and a channel though.
As a "for instance": for after hours emergency calls, I have two extensions set up. One with my cell number, and one with my wifes cell number. If someone calls and has an emergency, the * server rings our cell phones just like regular extensions. The call is using minutes and 3 channels, but my customers can reach me.
--
...because I care.

reply to wifi4milez

reply to nunya

reply to A_VoIPer

If your PAP2 is V1 you can load the firmware for SPA1001 and you'll have 2 lines (1 for Asterisk and the other for Broadvoice) using just 1 phone plugged in the port 1.

reply to wifi4milez