Asterisk Cookbook: Solutions to Everyday Telephony Problems
Apparently he's "been reading some interesting press releases about the company and wants to talk with me more about the phone situation here. There are many Asterisk "vendors" out their that can promise a lot in a hosted or premise system. Asterisk has arguably the most tested SIP stack in use today.
For us that is very valuable; however if you are going to just put an Asterisk on a public IP with Linux firewall and expect it to be enterprise grade, you probably have no business touching VoIP until you get some more experience. Again, we get back to Asterisk as a Part of the solution. In my opinion, in the enterprise or carrier world, it is a feature server.
While it has some tools to help with NAT traversal and other SIP signaling issues, if you are aiming to use those in hgh availability production think again. This is the job of a Session Border Controller. A real SBC in the provider core combined with Asterisk is highly reliable and robust.
This includes IVR, efax, conference bridging, etc. We have large financial clients that depend on high availability and large conference bridges.
We can accommodate participant bridges without issue. And regarding SBC you don't have to shell out Acme packet money. Edgewater's EdgeProtect, while not perfect, does the job in the hands of good engineers. Mind you we aren't running all the features on one big asterisk server.
auto-answering an originated call, dialplan / manager interface problem
This again is about understanding design and core roles. Try to run IVR, ACD, recording, conference bridging, voicemail and regular calling on one box, even with gold-plated hw specs at any scale and you'll see the issues. But build a feature server based architecture and you'll be quite pleased. Others have brought up the issue of threading in asterisk. For this very reason, we overbuild on RAM. Build your machine to do it's features entirely in the RAM space. If you get into the swap, this is where issues will come into play.
RAM is cheap. We don't have any feature server with less than 20Gig. So, the moral of the story is: know what you are doing. If you design as an engineer Asterisk can be an excellent component in high availability systems. The CEO of Junction www. If you can answer why a provider using Asterisk as the peer "proxy" is not 'getting it', well then you are probably well on your way to understanding the difference between well engineered solutions and providers just playing at VoIP. If you want to use it in low-use production or as a gateway, go for it.
But like I posted earlier, Trixbox has problems. I know i gush about the product but it really is that good. I've had my asterisk set up by a pro asterisk guy about 6 years ago. Were it not for power outages it would never go down. I've been very happy with it. Cost me about 5 thousand, and that was for the server and config. Switches to battery and back would cause the PS in the server to shut off. I was unaware of that issue, and called Trixbox support to help. Even though I was long out of contract, and not even upgradeable to the latest version Trixbox still helped me figure it out.
Totally nice of them and not the fault of trix or Asterisk. Don't be afraid of Asterisk at all. It's a marvellous thing. Do not take the Windows approach and just try to click your way through it till it works. Have one of the myriad Asterisk gurus set it up for you. And it won't cost you an extra grand or two for the remote access card like it does with other digital PBXs.
As for the development schedule of Trixbox, I have to laugh at the no new update since comment. I'll bet you can easily find digital PBX systems that have not had meaningful updates in more than one year too. Bug fixes yes, that's a given. I have been using Switchvox, which is one of the packaged and supported versions of Asterisk for some years now and have never had a lock up.
I have also never had a problem that couldn't be solved with tech support's help in less time than it takes to get a Cisco support tech on the phone. The hiccups have been few and very minor. Quality of call status and monitoring. Call recording, sharing and discussion. Automatic arhiving. Dashboards with top callers, company missed calls, daily traffic and more. Installation support included please contact! Manage Call monitor. User directory. Partner calls.
WEB Callback Website callback form. Analize Call history. Pivot view will give you an advanced call reporting tool. Instead of just having eth0, I have an OVS bridge named breth0 and eth0 is attached to that bridge. With this setup in place, I generated calls, which means both asterisk1 and asterisk2 have active channels. On asterisk Why channels? It also generates enough load on asterisk2 for the demo without making my laptop melt. What if in the simple forwarding case we could push this forwarding down into the kernel? To pull this off, first I needed to know about all of the RTP streams active on asterisk2.
I actually need to know about pairs of RTP streams.
- Asterisk Hacking with trixbox 2.6;
- Asterisk Call Bridging.
- Divertimento No. 2 in D Major, K131 (Full Score).
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Asterisk honestly does not make it very easy to get this information. I probably could have written an AMI script to get the info I needed. All of that sounded like too much work for my Friday afternoon hack. The easiest way for me was to write a custom Asterisk C module that provided a CLI command to dump all of the info I wanted. The output looks something like this:.
We do that using the OpenFlow protocol. OpenFlow lets you define a multi-stage packet processing pipeline. Each stage is a table.
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- Asterisk: The Future of Telephony - PDF Free Download.
Processing starts in table 0. Processing may continue in other tables based on what actions are executed. Each flow in a table has a priority. The flow that gets executed in a table is the one with the highest priority that matches the packet. If multiple flows at the same priority match, which one gets executed is undefined. What we want are flows that match an incoming RTP stream.
When a packet matches one of our flows, we execute these actions: change the source and destination MAC addresses, change the source and destination IP addresses, change the source and destination UDP port numbers, and send the packet back out where it came from eth0. Of course, typing up of those would be pretty tiring, so I just scripted it.
Here is a simple Python script to generate all of the flows we need:. If we want to clear all of these flows and let RTP go back through Asterisk in userspace, we can run this script:. This was just the result of an afternoon hack. My primary goal was just to spur some interest in exploring how cool things happening in the SDN space could provide new ways of doing things. If someone wanted to explore doing this in Asterisk more seriously, you could write some code in Asterisk that could speak OpenFlow to the local OVS bridge to create and delete flows as needed.
You could also imagine the possibility of speaking OpenFlow to a top-of-rack switch to push the forwarding out of the host completely, yet still through a controlled point in your network. If one end started sending garbage, this setup would happily forward it along.
To everyone, I hope you found this interesting and that it inspires you to go off and learn more about this cool technology! I began working on the Asterisk project in Thank you all for making the past seven years so memorable. Leif Madsen and I are working on a new book, the Asterisk Cookbook. One of the recipes that I am working on this morning is a method of adding debug statements into the Asterisk dialplan. I came up with a GoSub routine that can log messages based on log level settings that are global, per-device, or per-channel.