darmock
PIAF Developer
- Joined
- Oct 18, 2007
- Messages
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install-silk is now available for testing.
What does it do?
This program will install a precompiled binary supplied by digium onto your PBX in a Flash system.
What are the requirements?
Asterisk 10.X, Freepbx 2.9 and above, Centos 6.2 running either 32 or 64 bit. This codec ONLY RUNS ON ASTERISK 10! This is a digium decision so if you want it for Asterisk 1.8 please whine to them and not us. We tested this and while it should work with freepbx 2.8 odd things happened during testing so it was decided to not install it with freepbx 2.8.
Further info is here
https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats
We install the correct GENERIC codec onto your system. The automated processor id module that would install an optimized version is still not stable enough to let out of beta.
That being said it copies all of the 32 or 64 bit modules to /usr/src/silk32 or /usr/src/silk64 depending on your os. You can experiment to your hearts content and install the optimized one for your processor if you like. If your machine stops working.... well remember the arrows in the back paradigm. All instructions are on the web link.
We have also created a silk.conf file which gets copied to /etc/asterisk which make need to be edited. There is also a sip.conf sample showing how silk MIGHT be implemented.
We can provide NO SUPPORT of the silk codec as the PIAF Dev Team's involvement was to create a simple install program to get people experimenting with it.
How do I get it?
1. update-programs
2. install-silk
or
2. pbx-menu then choose install-silk
Just as an aside digium reports that the silk codec can take up to 3 times the processor resources that g729 does.... Good luck with an underpowered server.
Tom
What does it do?
This program will install a precompiled binary supplied by digium onto your PBX in a Flash system.
What are the requirements?
Asterisk 10.X, Freepbx 2.9 and above, Centos 6.2 running either 32 or 64 bit. This codec ONLY RUNS ON ASTERISK 10! This is a digium decision so if you want it for Asterisk 1.8 please whine to them and not us. We tested this and while it should work with freepbx 2.8 odd things happened during testing so it was decided to not install it with freepbx 2.8.
Further info is here
https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats
We install the correct GENERIC codec onto your system. The automated processor id module that would install an optimized version is still not stable enough to let out of beta.
That being said it copies all of the 32 or 64 bit modules to /usr/src/silk32 or /usr/src/silk64 depending on your os. You can experiment to your hearts content and install the optimized one for your processor if you like. If your machine stops working.... well remember the arrows in the back paradigm. All instructions are on the web link.
We have also created a silk.conf file which gets copied to /etc/asterisk which make need to be edited. There is also a sip.conf sample showing how silk MIGHT be implemented.
We can provide NO SUPPORT of the silk codec as the PIAF Dev Team's involvement was to create a simple install program to get people experimenting with it.
How do I get it?
1. update-programs
2. install-silk
or
2. pbx-menu then choose install-silk
Just as an aside digium reports that the silk codec can take up to 3 times the processor resources that g729 does.... Good luck with an underpowered server.
Tom
