afrimesh-potato_r635-0.4alpha-1
Antoine van Gelder
Two gifts for this most merry of Dies Natalis Solis Invicti.
#1 - EXPLICIT SERVICE RELOADING
Changing the potato settings now results in a reload button popping up next
to the appropriate section. Pressing the button will reload the specific
service and apply the settings to the potato.
Assuming you are connected to the potato via the LAN port there are no
settings that can be changed which will render the potato inaccessible.
If you are going in via the Mesh it's a different story though.
Thoughts would be muchly appreciated on the seriousness and/or mitigation
of this risk.
#2 - TELEPHONY CONFIGURATION
Given that, socially, it's a lot easier to add functionality than to take it
away I ended up optinf to start with a simpler SIP config than the one
laid out by David in his 'Challenges' email.
In the process I ended up learning far more about Asterisk than I ever intended but did eventually manage to persuade my Potato to act as a
SIP client.
Step 0: Update VT sources, rebuild & reinstall firmware
Replace /etc/config/asterisk & /etc/init.d/asterisk with the patched
versions I sent to the VT list earlier.
Step 1: Configure a SIP user on the IP04 VoIPtel GUI e.g.
Extension: 7001
Name: Mesh Potato
Password: mysecret
Dial Plan: (optional)
SIP: Enabled
IAX: Disabled
NAT: Enabled
Step 2: Then on the Mesh Potato GUI set as follows:
Trunk SIP : enable
Address : address of IP04
Username : same as 'Extension' in IP04 config
Password : same as 'Password' in IP04 config
Step 3: Test
. Four digit extensions should give you the extensions setup on the IP04.
. Dialing 0 should give you outside line on IP04 if you have one configured.
. Dialing 7001 from the IP04 should ring the MP
Using this setup in my office I can now use the MP to:
. Dial the other MP's on my testbed via IP address
. Dial the POTS handset on my desk which is hanging off the IP04
. Access my PSTN outside line which is plugged into the IP04
. Phone my wife's POTS handset which is hanging off a Linksys ATA
hanging off a Meraki mesh router hanging off my testbed
. Dial the SIP phone on my laptop
Of course I can also dial into the MP from any of the above.
I've now got quite flexible control of Asterisk config from within Afrimesh
so moving forward incrementally from this point into more complex
configuration territory should go relatively quickly.
--
...aaaaaand that's the last useful thing anyone's going to get out of me until I've had a serious break.
I'll be back in action come Monday, January 11.
Geseende kersfees en voorspoedige nuwe jaar almal! :-D
- antoine
--
http://7degrees.co.za
"Libré software for human education."
David Rowe
Something that just occurred to me.
Asterisk has a web server with a AJAX API called AJAM built in:
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk
+Manager+(AJAM)
The AsteriskNow GUI is based on AJAM (the Voiptel GUI is a modified
version of AsteriskNow).
AJAM may be useful for Afrimesh integration, for example on a MP you can
use it to poll the status of SIP and IAX trunks as an alternative to
'asterisk -rx' shell calls.
Not sure if there is any clear advantage in AJAM versus conf file
methods. I think you still need conf files to set up sip trunks and
extensions.
Cheers,
David