They’re running SP2 on an Aastra system, but although we’ve not tested with Syspine and D-Link we’re pretty sure this will happen with those systems as well.

The customer uses analog trunks with a hunt group, and when one of their people called their published number from the inside (with a ‘9′ of course) not only did the  call not complete, but another existing conversation with an outside party got dropped!  The customer was able to replicate this problem.

So (for now at least) don’t go calling your external number(s) from the inside!!

Has anyone noticed that you can configure two VoIP Providers under SP2?

(Example: Dial a “9″ + number for the configured #1 NGT Service and dial a “8″ + number for the configured #2 Packet8 Service)?

 
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• Do a backup.

• Find and open the corresponding settings.xml (e.g. if your lab phone belongs to user 100, find the file under XXXXX backup @ 2009-XX-XX, XX.XX.XX\Contacts\100\settings.xml

• Find the corresponding call forwarding rule section(e.g.
<callhandling>
  <rule>
   <step timeout=”20” callusing=”internal”>vm</step>
  </rule>
</callhandling>

Change 20 to any number, can be 60(1min), 120(2mins) etc..

• Save the file

• Do a restore

Please note this is just a trick (not documented and not an official solution). 

After you change the value, any change through the UI for this extension will cause the value be reset. So either you don’t change that extension afterwards, or you’ll have to repeat the steps described above again (do a backup/modify timeout/restore).

When the customer calls in, if they press “#” and wait a couple of seconds, they can then speak the name of their destination as normal.

This is certainly less useful than the previous behavious, but it’s better than nothing.

Hope this helps!

 How to recover the phones in web recovery mode:

1.  Phones with web recovery display an IP address on the LCD.
2. Connect to that IP address using a web browser like IE or Firefox.
3. Grab the firmware file of the phone you want to put on the phone.  For example for 6753i RP it would be a file with “6753irp.st” name.
4. If you have a TFTP server with your network you could use that.
      Otherwise you could download PumpKin from internet (freeware). See below about how to use PumpKin.
5. Place the phone’s firmware file in your TFTP’s root path.
6. On the web recovery page of the phone, specify the “server IP” as the TFTP server IP address, the firmware file name as the name of firmware you want to upload to the phone (i.e. 6753irp.st) and choose the protocol as TFTP.
7. Upload the firmware file and the phone should come back on.

A Quick reference for using PumpKIN TFTP server

TFTP Server Set-up

There are a number of TFTP servers available. PumpKIN is one of such TFTP servers. Use the keywords “pumpkin TFTP server” on Google and you should get the web site where you can download the software
from. Installing PumpKIN is straightforward. To configure the directory from where you would be serving the files, click on the Options button on PumpKIN’s main window as shown in the following figure.

PumpKIN Options

It is important to select the “Give all files” radio button under the “Read Request Behavior” category. This makes the files to be served without any manual intervention when requested.

If you want to prevent users from writing files to the directory select the “Deny all requests” in the “Write Request Behavior” category. Click the OK button after you have entered all the required information. All the firmware files should be in the file system root directory. Currently we do not support downloads from files present in sub-directories. Consult PumpKIN’s documentation if you need more information on how to set-up the TFTP server.

Command Codes for Syspine
Code Function

#*IP(47) Gives phone IP address in the display

#*MENU (6368) Shows phone menu in phone display (Adjust
ringing here and much more)
#=Enter
Call History Key takes you back one level
Lift and replace handset to exit phone menu
UP/DOWN arrows to scroll

872# Code to access external paging amplifier

822# Access to auto attendant

886# Access to Voice Mail

8* <slot> # = Retrieve parked call (not documented on help screen)
*08# = DND Toggle
Speaker Phone = alternate exit from Menu
Mute = Clears out digits on some setting screens

Suggestion: Default ‘Hold Recall’ timer appears to be 180 seconds,
but the phone drops the call after 120, so you don’t get any
warning. To correct, change ‘Hold Recall’ under ‘Preferences’ to 30
or 60 seconds. This can also be done from the web interface.

We also checked this approach with the tech dudes on the Microsoft Response Point team.  They were comfortable with this approach, so I’d be comfortable recommending this to other customers in the future.

Bear in mind that any phone-specific customization will not be carried over – it’s just the Response Point config.

What is required is the MAC address of the base unit. The MAC address is used so that when you provision NGT service, the system can pull the correct configuration from NGT’s database. So, if you change your base unit for some reason, be sure to let the good folks at NGT know about the new MAC address prior to configuring NGT service.

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.

Scenario:

A Syspine Response Point system configured with two IP Phone 310 handsets, our target handsets is setup as extension 102. Since there’s not a way to get the IP address of the phone directly off of it’s LCD display I just looked it up on my router, a Kentrox Q2300 which is acting as my DHCP server. By comparing the MAC address on the back of the phone in the list of DHCP leases on the router we see that the phone we want to dial has an IP address of 172.16.83.104.

Now that we know the extension number and IP address of our target, we can browse to our Snom and make our call!

Notice that before the IP address I placed “102@” designating the extension which I was dialing into on the Syspine 310 handset. In the scenario we just did, this is optional; in fact any arbitrary number can be used or it can even be left out completely and the 310 will still ring. Adding the extension would be useful however when a target handset has multiple extensions on it and you want your call to be able to roll into the proper voice-mailbox if the call is not answered.

As part of working through this article I even moved the Snom to several different IP networks in my lab and it continued to work even across the router. This is significant as it offers up some openings from the typical Response Point requirement that the handsets must be on the same subnet as the base unit.

John