According 'a option listed twice in configuration file' issue ...
Imagine following configuration file:
<flat-profile>
<Resync_On_Reset ua="na">Yes</Resync_On_Reset>
<Enable_CDP ua="na">No</Enable_CDP>
<Enable_CDP ua="na">Yes</Enable_CDP>
</flat-profile>
There are two instances of Enable_CDP. So we can be sure at least second one is considered change of the value - regardless the original value of option.
Any change of Enable_CDP will trigger phone reset, thus phone resets whenever this configuration become loaded.
Now how about Resync_on_Reset. The 'Yes' value mean the provisioning will be reloaded on bootup. So phone enters never ending loop:
- Phone loads configuration, the reset is committed because of it
- Phone boot up, then (re-)loads configuration
- continue with step 1
But at least "Info/" entities can be safely dropped, right?
True.
can we put more stuff into spa504G.cfg? Like those regional settings, etc? And remove them from individual
Phone process all configuration files the same way. For the purpose of this discussion you can assume the phone will concatenate all of them into one file, then parse it. The real algorithm is slightly different, but you can ignore the details now.
For example I'm using just simple configuration file: <Profile_Rule>https://$(A)/Cisco/Provisioning.php</Profile_Rule>
The Provisioning.php script on server will merge all necessary stuff for particular phone and send them to phone in one chunk.
Do you care to share Asterisk-specific options you use?
There is just only true Asterisk-specific option I know: <Server_Type>Asterisk</Server_Type>
Other are not specific to Asterisk, Yes, I'm willing to speak about any option or to share complete configuration, if you wish.
Lets go.
IP setup
You can use either static or dynamic configuration. But first I need to explain the terms.
Saying 'dynamic' I mean there's a pool of IP addresses. DHCP server will assign a rantom address from the pool to the phone.
Saying 'static' I mean the particular phone will receive same IP address all the times. Such address cam be assigned by DHCP to them (but not randomly from pool) or hardwired by phone configuration.
So 'static' is not the same as 'with no DHCP'.
With dynamic configuration the Asterisk know no IP of phone unless phone become registered.
With static configuration you may configure no registration on phone (IP address of phone must be filled in Asterisk's sip.conf) or you may register even in such case.
The best choice depends on local condition. I would like to recommend you static configuration with DHCP with registration. E.g. phone is configured to use DHCP. DHCP server use no random address of pool, but assign preallocated IP according phone's MAC address. There are no IP addresses in sip.conf, phone will register to Asterisk.
I dislike the dynamic (random) addresses. it make debugging hard and old logs may be useless (as you doesn't know what phone belong to what IP address).
Your mileage may vary.
RTP Packet Size
It's not Asterisk related option. You should follow you upstream Voice provider recommendations here. E.g. you should use the same value he is using.
0.020 is most common value used on G.711 - just because it's standard value for classic (e.g. non VoIP) phone telecommunication world. As long as you will use G.711 we can assume the 0.020 is correct. But check parameters of incoming INVITE (from upstream to Asterisk) - the ptime value of SDP will disclose provider's choice to you.
Reorder Tone
It has nothing to do with Asterisk. It's the definition of tone generated locally by phone when an outbound call has failed or when a person on the other end hangs up after a call has been established. It's used also when a dial tone or any of the other options times out. Reorder tone may be replaced by SITx tone when requested by <SITx_RSC> option.The definition of reorder tome should follow regional requirements (they are country specific, for exact advice I need to know the country the phone is located in).
Daylight Saving Time Rule
Have nothing to do with Asterisk either. it need to follow your country daylight saving rules.
NAT Keep Alive Intvl
... also have nothing to do with Asterisk. If there's NAT between phone and Asterisk, it needs to be set. Interval should be lower than NAT purging time. If there's no NAT you can leave it as is (superfluous keep-alives harm nothing) or set it to 0.
Multicast Address
Such address is required if there's Linksys call control device. Asterisk can't emulate it. But if you will use multicast paging feature on Asterisk, this address needs to be set to the same value you configured on Asterisk. I would like to recommend you not to use multicast paging. It have more drawbacks than positives.
Sorry for many questions, trying to understand process better
That's we are here for. Just rate valuable comments ;-)
