RP Tricks

Asterisk and Syspine Interconnectivity

The following article was sent to me by one of our readers, Farhan Sabir, a systems engineer at www.cigear.com up north in Ontario, Canada.  If any one else wants to submit an article for publication please let us know! We welcome contributions.

John

Scenario

Syspine Digital Operator Panel (DOS-A50) with 4/8-port FXO and Syspine IP Phone 310. Targeted Extension 101.

Asterisk Server with Polycom phones, extension 255 working on Polycom SoundPoint IP 501.

First, lets check the IP of syspine extension 101 by dialing #*47# gives 192.168.1.117 (example only)

Polycom 501 ext 255 on Static IP: 192.168.2.205 (you can find it out from MENU-2-2-1)

Dialing from Syspine IP310 to Polycom:

Dial: *205#
This actually dials 192.168.1.205 which is Polycom with Ext 255 registered to Asterisk.

Dialling from polycom to ip310:

Instead of trying to dial from the phone, I configured an extension on Asterisk with number 101 (same as syspine). You just need to mention it has the same ip as IP310 i.e, 192.168.1.117 in this case. Now, from polycom I dial 101 and rings the syspine ext 101.

Note on DHCP/Static IP

Syspine base unit and the IP-310 phones have to be on DHCP to work properly. Most of us on the other hand, want to use static IPs. The only solution I see to it is having a better DHCP server that can assign IPs based on the MAC address. This way your phone always gets the same IP, making it kind of static.

Additionally:

1 – SIP Trunks (Pre-SP1): you can use external ATAs to use your sip trunks and hook them up to the Syspine ATA (PSTN ports). Until of course you can get SP1.

2 – SP1 allows you one SIP Trunk. Again, if you need more than 1, you need the FXS ATAs. Or you can find a provider with multiple channels on one single SIP trunk. The one I have right now gives me 10 channels. Normally its 2 channels per connection. (Not sure if syspine can use multiple channels).

3 – If you only want to see how syspine works, you need to either hook up your office lines to it, or find another way around. Using one of your office lines for testing syspine may not be a good idea as you don’t want to tell your existing customers that your main line id being used for testing syspine for a few days. So, once again, I create Asterisk SIP extensions and configure them on external ATAs. Take this phone lines and hook them up into syspine as PSTN lines. Its just that they aren’t the actual PSTN lines, but extensions of another ip pbx. But this makes my life easy.

4 – MSRP has the sip trunk limitation, I like to have more than one just to keep testing different things. So… I use asterisk with ATAs or with MSRP SP1 and dial as many trunks as possible.