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Asterisk / FreePBX SIP Trunk Settings for Phone Power

To much work for me....

That is why I would pay a few extra buck and let PP do the work for me

»www.phonepower.com/services/voip···essPlans
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I got the need for speed »bit.ly/topgunparody

reply to tevet

Thanks for taking the time for me, but a few questions. I have to proceed carefully since if there are too many SIP Agents on the account it will refuse to register no matter what unless I call in and ask them to clear the Agents.

port=12060 Is this entry for the proxy address? I put my PP credentials into a SPA2102 and it was working fine with the line for the proxy as 208.64.8.6:12060.
And the accounts both worked on the spa with 5060 and 5061, that is why I was wondering how to assign that in freepbx/pbxinaflash

register=6185551212:LONGPASSWORD1@208.64.8.6/Phonepower1

Is this equivalent to the line in pbxinaflash on the trunk page at the bottom where I put the register string?
Also, in this example, what is Phonepower1?

PhonePower states they want the registration interval to be between 1 and 4 hours. What entry will control that?

I believe that I may have this registration issue too. It worked for a while, but this morning, I had no service, incoming or outgoing with PP via Asterisk.

reply to lineofsight
First, let me warn you that I'm not currently running my own Asterisk system -- I have an account at PBXes.com, which is Asterisk based, is working properly with PP and I'm just reporting the setup that their interface generates.

If you attempt more concurrent registrations on PP than allowed, the excess will be rejected with 403 Forbidden. However, if you are not actively refreshing them, the old ones will eventually expire and the new ones will be accepted. This is what happens, for example, if (because of a lost connection or ISP policy) your public IP address changes. This sort of thing happens fairly often and recovery should certainly not require manual assistance from PP. When registration has been refused, have you simply tried waiting an hour? Of course, for this reason, you probably don't want to use an expiry longer than an hour.

There is a setting defaultexpiry that controls the requested registration interval. However, the default is short (I believe 120 seconds) and when PP sees that it should return an error demanding at least 3600, and Asterisk should comply. So, I don't think that setting defaultexpiry should be necessary.

The /Phonepower1 at the end of the register string is an extension name (you could also use a number). When Asterisk gets an incoming call, it acts as if that extension was dialed. That lets you handle calls differently, according to which account they came in on. If you don't put anything there, Asterisk starts at extension "s". With some providers (though IMO not PP), it's required to put your user ID in there, e.g.
6185551212:LONGPASSWORD1@208.64.8.6/6185551212

Sorry, I know nothing about PIAF.

reply to lineofsight
On Asterisk, bindport= specifies the local port number (the port it listens on and the source port for outbound requests); port= specifies the port on the peer's server. As far as PP is concerned, bindport can be anything, though your extensions must be configured to connect to Asterisk on whatever port is used for bindport.

On the SPA, SIP Port specifies the local port (the port it listens on for that line). Unlike Asterisk, you can use different values for the two lines. They can usually be anything, though with some routers you need to have them unique across all devices on the LAN. The server port (if other than 5060) is specified in the Proxy or Outbound Proxy field with a colon followed by the port number.

reply to tevet
Hey guys, due to the fact we're seeing threads similar to this pop up in forums all over the web, we'll be putting up an article in our Knowledge Base this week with some basic asterisk settings for the SIP Credentials. Be aware, it's still play-at-your-own-risk and our Support dept won't be able to help you set up your asterisk box with SIP Creds.

To be honest, the SIP Credentials feature was launched to satisfy our residential/SOHO users who wanted to use their own unsupported softphones, mobile apps, etc. It wasn't really envisioned for asterisk users, since we already have a fully supported SIP Trunk product that supports asterisk.

So far, our SIP Trunk product has done pretty well with minimal marketing effort behind it. There's a fairly comprehensive interop process that eliminates most issues right out of the gate. If you are interested in getting more info, trunking packages start at $55. Contact sales@phonepower.com for details on the packages.

As mentioned, the KB article is being created, but here's our recommended settings for those interested:

[general]
;There will be other bits here but the following value should be changed
defaultexpirey=3600

; This will register your line to PhonePower and make it available via extensions.conf as [[SIP User ID]]
register => [[Auth ID]] [SIP Password]]@208.64.8.6:5060/[[SIP User ID]]

;This defines the peer.
; Notes are included to aid understanding
[phonepower-sip]
type=peer ;peer is used because it is a bi-directional channel
context=from-trunk ;context for calls originating here
insecure=very ;If this is not set inbound calls will not work
nat=never ;Our border elements will handle this. Configuring NAT traversal will break more than it fixes
dtmfmode=inband ;In our experience in-band DTMF with asterisk was much more reliable than RFC2833
username=[[SIP User ID]]
secret=[[SIP Password]]
authuser=[[Auth ID]]
host=208.64.8.6
fromuser=[[SIP User ID]]
fromdomain=208.64.8.6
maxexpirey=3600 ;
minexpirey=30 ;
disallow=all ;These 3 lines ensure all calls will only use G711 or G729
allow=uLaw ;
allow=g729 ;

It is strongly recommended that IP tables be configured as well to prevent unauthorized access.

The following is a rudimentary firewall config for an Asterisk server with a single network interface. As always this is for expert users only.

iptables -A INPUT -i lo -j ACCEPT
iptables -A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT
iptables -A INPUT –p tcp –dport 22 -j ACCEPT
iptables -A INPUT -s 192.168.0.0/16 -j ACCEPT
iptables -A INPUT –s 172.16.0.0/12 -j ACCEPT
iptables -A INPUT -s 10.0.0.0/8 -j ACCEPT
iptables -A INPUT -s 208.64.8.6 -j ACCEPT
iptables -A INPUT -j DROP

You will need to follow the directions for your Linux distribution to make this firewall persist on reboot.
For Redhat based distributions “/sbin/service iptables save” will save this config and “/sbin/chkconfig --level 345 iptables on” will ensure iptables starts with this config after each reboot.

Update: I received info from PP that ports 5060 and 12060 now behave identically as far as the server is concerned. Port 12060 is used to avoid trouble with consumer routers that have problematic SIP ALGs (which normally monitor port 5060). If you don't have such an issue, registering to port 5060 should be fine, though you still may want to avoid bindport=5060 for security reasons.

reply to PhonePower
Great info.
I have 2 accounts, that's 2 Auth ID and 2 sip passwords, 2 Sip User Id's.
Can't get them both to register. Tried 5061 in the registration string instead of 5060, it won't register.

Both are registerd now. One on 5060, the other on 12060

Nope. Spoke too soon. They both deregistered when I tried to call in to those numbers from my cell.
I can get one to register, it will call out. Will not get incoming calls.

Thanks for everyones help.
Found some issues with the instructions here from PhonePower guy.
Defaultexpirey, Minexpirey, Maxexpirey are misspelled. It is supposed to be "expiry" no "e".
Also I deleted the comments from the example and the trunk works for inbound and outbound calls.
Still can't get the second trunk to work at all.

You are correct that 'expiry' is the right spelling. However, the author misspelled it when he first wrote the code, and by the time he learned about the mistake, there were lots of config files depending on it. So, he modified the code to allow either; see »www.mail-archive.com/svn-commits···594.html

Unfortunately, that doesn't explain why you are having the registration problem. Conceivably, there is a server side issue with multiple accounts registered from the same bind port. More likely, if Asterisk is not on a public IP, your router might not like two destinations with the same source port. (Port forwarding may help, but that's a security issue, especially if your bindport is 5060). You might see whether having only the second trunk works ok. Can you capture traffic with Wireshark to determine what is going wrong?

I have both accounts/trunks registered. I can get outbound to work on both of them.
Incoming on the first trunk works consistently.
Not on the second trunk. Sometimes it works inbound, a few minutes later, nothing comes inbound on that trunk.
And the inbound calls show on the first trunk even if the call was inbound on the second trunk number.
This is where I think a different port number from PP would make a difference. 12060 and 5060 act identical.
I quickly check the registrations and it always shows registered on both trunks.
The interesting thing is that I have been using the Grandstream Ht-502 supplied by PP and it has no issues with my router; both accounts are on that single unit with acct 1 on line port 1, acct 2 on line port 2.
This means it should be able to work since the router can sort that out correctly.
I can't do wireshark as I would need a PC between the router and the device, I don't have that capability.

reply to tevet