SPA-2000 Set-up Problems

[JamesLewis]

Hi there

I have recently bought two SPA-2000 units, which I want to use alongside my Grandstream Budgetone phone, which is already working fine.

I have a Linksys BEFSX41 router, so the phones will be using NAT.

I assume that the Budgetone is using the Stun server, as when I installed it, the phone didn't work until I entered the Stun server setting.

Please can anyone help me configure the SPA-2000 units to work on the same network with the Budgetone. I can make the budgetone ring but there is no audio between, and I can ring the SPA-2000 from a land line but theres no audio coming back to the SPA-2000.

I have this info from Pipecall who are my SIP provider, but they say they don't support the SPA-2000

SIP Server: sipproxy.pipecall.com
Outgoing Proxy: sipproxy.pipecall.com
STUN server: stunserver0.pipecall.com:10000

When I rang the support line, the guy said something about changing ports to get it all to work on the same LAN...?

Well, any help would be *greatly* appreciated.

Best regards
James

[mberlant]

James,

Several things you mention here give me the shivers. High up on the list is the suggestion about "changing ports to get it all to work." This from a company who has their STUN server on a port other than 3478.

Right now, I am in my home/office. I also have a BEFSX41 here, with an SPA-2000, an SPA-3000, a Grandstream 486, and X-Lite on two different laptops. All coexist peaceably with no ports forwarded manually through the router.

Despite Pipecall's help desk giving you incomplete or incorrect information, I think I found enough pieces to get you going. (And they have the nerve to charge 0871 rates for this "help"?)

Proxy: sipproxy.pipecall.com
Outbound Proxy: sipproxy.pipecall.com
Use Outbound Proxy: yes (you may have to play with this)
Use OB Proxy in Dialog: no (you may have to play with this)
User ID: Your entire phone number 08458xxxxxx
Password: Your pipecall password
Auth ID: sipCCCCCCCC (where CCCCCCCC is your assigned Authentication Username)
Use Auth ID: yes

STUN server: stun.softjoys.com:3478 (I don't trust a STUN server that operates on a different port than 3478.)

Set up each SPA-2000 one line a time, testing Line 1 before configuring Line 2 and then adding the second SPA-2000 in the same manner. I recommend that you run the Voxilla wizard (FWD with STUN) to put most of the bits into alignment before manually poking in your pipecall credentials.

Good luck, and let us know how you make out.

Michael

[JamesLewis]

Thanks for your reply and your help, Michael
I will try those settings tonight when I am home.
Are you able to call using more than one phone/unit simultaneously on the same network, and are you able to call the other phones on your network and speak to and hear the other phone ok?

[mberlant]

Yes. It depends.

Yes, you can have multiple simultaneous conversations flowing through the same router and DSL/cable pipe. If you are using G.711 CODEC each conversation will require 80kbps of bandwidth in each direction. On my 8M/1M ADSL pipe I have had as many as three simultaneous connections (haven't tried for more) plus P2P, etc., with no voice degradation in any conversation. If your ATA and PipeCall have a low-bandwidth CODEC in common you can use that instead. The tradeoff when using compression is that you give up some voice quality and latency in exchange for a smaller data package. The degradation can range from imperceptible to negligible to tolerable to awful - it's up to your ears and the karmas of technology to see how you will make out.

It depends on your router and your service provider as to whether a "hairpin" connection will work for you. As you can tell from the term, hairpin refers to a call where the originator sends out the call request from one LAN only to have it completed to a destination on the same LAN.

Some service providers, like FWD, connect calls by telling each device of the public parameters of the opposite party and ordering them to set up a direct voice path. Some other service providers, like Vonage, route all voice traffic (along with control traffic) through their own switches, so each half of the conversation is an independent connection.

The Linksys router does not permit hairpinning for impermanent connections, so a FWD call from one phone to another inside the same LAN will connect but will not establish a voice path unless you fix the LAN IP address, SIP Port and RTP Port Range of each ATA and forward all of them permanently through your router.

I have no idea how PipeCall treats on-net connections, nor how serious you are about making intra-LAN calls, so you will need to test this for yourself and decide from there.

[JamesLewis]

Hi there, many thanks for your help. I set up as you suggested and the lines at working and can all make and receive calls. As you say if I call another phone on the same lan they ring but you can't hear. It's not too much of a problem but I would be interested to know what it would require to set them up so they can call each other?
Thanks for your help!
James

[mberlant]

As I said above,

The Linksys router does not permit hairpinning for impermanent connections, so a FWD call from one phone to another inside the same LAN will connect but will not establish a voice path unless you fix the LAN IP address, SIP Port and RTP Port Range of each ATA and forward all of them permanently through your router.

It's a bit of labor to accomplish this, but it should do the trick.