
We've deployed quite a few Draytek 2820's over the last year or so along with Asterisk PBX's.
What we found with a significant number of installs, is that every few days, Asterisk would report the termination servers as being unreachable. We explored Asterisk at first, but we then found a reboot of the modem would clear the issue.
There comes a point where Asterisk, great as it is, starts to hit user limits. While asterisk 1.8 is said to handle larger loads, it's simply not mature enough for me for a mission critical platform. The consensus seems to be that with 1.4 at least, 200 phones (SIP endpoints) is about the maximum. Opensips on the other hand, boasts 2,000. So rather than wait until the platform falls over, I'm kicking off some Opensips/asterisk development.
Read more: Installing OpenSips, Asterisk and A2Billing on CentOS 6
At Noble Solutions we have a pretty large amount of SIP peers and when we need to check them it can take a little while to look through them all using the asterisk command line interface. In our endless search for speed and efficiency we've come up with a quick and easy solution to help us check a single peer or a range of peers direct from the CentOS command line.
Read more: Check the Status of an Asterisk Peer from CentOS Command Line
At Noble Solutions we handle call recordings for a large number of clients and the number of files stacks into the thousands! We recently had to find a very specific list of files amongst this very large pile and needed a way to find specific files created on specific dates. Heres how:
Read more: Search CentOS for a File Created on a Specific Date.
One of the most requested functionality prior to Asterisk 1.2 was shared line appearance. It finally appeared in 1.2 but it's take-up has been somewhat muted. Part of the reason for this is that it uses conferencing and while it works as expected with Line appearance, you can't use the transfer buttons on the phones. In this article, Nik Middleton explores an alternative approach.
As mentioned in the intro, the embeded SLA functionality in Asterisk uses conferencing in order to achieve its goals. Conferencing is a drain on the CPU and in addition requires a timing source such as a digium card, or failing that the use of ZTDummy. In addition, because you are in a conference, you need to use the native Asterisk transfer functionality (normally set to # in features.conf)
So, given the above, it's not surprising that SLA hasn't had a high adoption rate, yes the lights flash for the appropriate lines, but you can't use your BLF or speeddials to transfer calls. Essentially all the fancy features on your phone are useless.

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