[asterisk-users] SPA-962 & Asterisk

[John Todd]

On Nov 4, 2008, at 6:08 AM, Steve Anness wrote:

> Good Day,
>
> I have been tasked with fixing the time on our asterisk server.  I  
> am having a hard time finding documentation to tell my what asterisk  
> uses to get its time information to push to phones (or a better  
> question, where does the SPA-962 get its time information)?
>
> Basically, I can go under the settings of the phone and change the  
> offset to set the correct hour, but it is still about 4 minutes  
> fast.  So the SPA-962 has an offset option, but to offset it from  
> what?  The time on the asterisk server?  That isn’t right because my  
> asterisk server has the correct time.  To offset from GMT?  No  
> because I am +6 from GMT not +2.
>
> I can physically set the time, but that is a bitch when you have  
> many phones, shouldn’t the phone be syncing with something?
>
> Any thoughts?  I am not finding anything conclusive.
>
>
> Steve Anness
> ICT Support Analyst
> Humanitarian International Services Group

Your SPA devices almost certainly get their timing data from an NTP  
server.  Some devices will find their NTP servers with DHCP options  
requests, and I think there is even a Zeroconf method for determining  
local NTP servers, but I doubt anyone uses that method.

One of the more novel methods I used a while ago (when I was doing  
design for ATAs) was to use the Date: header in the SIP INVITE as the  
time set.  The device would know it's GMT offset, and then calculate  
what time it was.  It was an ugly hack, and that ATA was the reason  
that the Date: header is in INVITEs in Asterisk.  :-)   I was kind of  
ashamed of it a while ago (NTP is much more correct solution) but the  
more pragmatic I get in my old age, the more elegant it seems.  No NTP  
stack to worry about, no additional firewall holes, etc. etc. and  
phones really don't need to be super-accurate so NTP is typically  
overkill anyway.  ATAs especially, since they don't even display the  
date - they just send it along with the Caller ID data when a new call  
event is generated.

JT


---
John Todd
jtodd at digium.com        +1-256-428-6083
Asterisk Open Source Community Director