When our process is complete, all billing, provisioning, termination, and call rating functionality offered with the version of AgileVoice chosen by our clients is ready for launch.
In addition, our installers are extremely competent with VoIP technology as well as Asterisk PBX and SER in specific, and the Operating Systems and surrounding performance, security, and scalable issues relevant to each implementation.
The following is an excerpt from an Installation/Integration follow up e-mail to one of our clients from an AgileVoice installer after their turnkey AgileVoice solution was complete:
I have installed the provisioning part of AV on the two Asterisk systems. It is set via cron to run every 15 minutes. The sample product and DID in the system did properly export. In addition, the few CDRs that were in the Asterisk logs did import.
Using the Digium provided source code will lose some AV functionality, such as conferencing and fax capabilities. In addition, the customizable HTML templates for voicemail emails and fax emails won't function. There are also stability patches included in our version that simply do not exist in Digium's version. Many of the patches we've applied are awaiting entry to Digium's source code but have not yet made it into their source code system.
In order for voicemails to be managed through the AV web interface, I need to set up NFS on one system. Which system is the primary system that will service everyone? Remember, Asterisk doesn't yet fully horizontally scale, thus you can not load-balance requests. Asterisk will vertically scale on Solaris.
Additional items:
- Using the from-pstn context is a good idea. Just be sure to include the default context, so that customers numbers are routable from the pstn. However; I would not include any additional contexts, except maybe didprepaid, etc. Contexts serve as part of the security of Asterisk.
- NTP in daemon mode should be configured on all systems. Without NTP, each system slowly drifts and can make for a hard time trying to debug issues across the multiple systems. In addition, Asterisk requires a good clock source.
- If you opt for firewall rules, make sure to leave all UDP traffic open or configure Asterisk to use a certain window of ports that you've opened.
- Asterisk should be ran in real-time process scheduling priority. However; it should be monitored to ensure it works as expected on your platform. To make this happen, Asterisk should be started with the -p flag. Again, carefully. Any process running real-time can freeze the system.
- All unnecessary services should be disabled on all systems. (eg: cups) There isn't a good reason for a production voip server to run a print spooler.
- X Windows should not be running on a production server. There are far too many exploits with X Windows.
- QoS should be employed on your network to ensure your voip traffic is held in high priority, least latency fashion.
- You should carefully run some testing on "phone1" It appears to be having some hardware issue that will probably interfere with voip. It may not appear on one or two calls, but will show up under some load. The information below was captured from "dmesg". It signals that the system is having a hard time with the IRQs. (Pay no attention to the NIC going up and down, unless you haven't been rebooting the box; otherwise, you have a switchport, wire, or something bad.)
Losing some ticks... checking if CPU frequency changed.
tg3: eth1: Link is down.
tg3: eth0: Link is down.
tg3: eth1: Link is up at 1000 Mbps, full duplex.
tg3: eth1: Flow control is on for TX and on for RX.
tg3: eth0: Link is up at 1000 Mbps, full duplex.
tg3: eth0: Flow control is on for TX and on for RX.
tg3: eth1: Link is down.
tg3: eth0: Link is down.
tg3: eth1: Link is up at 1000 Mbps, full duplex.
tg3: eth1: Flow control is on for TX and on for RX.
tg3: eth0: Link is up at 1000 Mbps, full duplex.
tg3: eth0: Flow control is on for TX and on for RX.
warning: many lost ticks.
Your time source seems to be instable or some driver is hogging interrupts rip default_idle+0x20/0x23
I like to warn our customers about these items, even though we do not offer or support the administration of systems or networks at this time. Many items lead to poor voip quality and any and every little thing contributes.
For your backup scripts, you'll want to ensure that at least /root and /etc/asterisk make it into your backup routines on the Asterisk systems. For the web server and database, you'll want to ensure you make MySQL backups and you'll want to at least backup the /var/www/html directory.
-Joe
AgileVoice.com
