Knowledge Base

The Digium Knowledge Base is a searchable library of troubleshooting advice and current information updated by Digium support technicians.
Search:     Advanced search
Browse by category:
Contact Us
Knowledge Base / Software / Can I use G.729 on a 64-bit system?

Can I use G.729 on a 64-bit system?
Views: 4953, Votes: 1

Posted:
15 Mar, 2006 - Klopfenstein C.

Updated:
03 Oct, 2006 - Support D.
Print Email article
The G.729 (Codecs/G.729) codec can be used on AMD64 based systems, but you will first need to edit the Makefile to have Asterisk compile with '-lz' to link it with the zlib library for compression. You should add "-lz" to the end of the "LIBS+=-lncurses -lm" line in the Asterisk Makefile. It should look like "LIBS+=-lncurses -lm -lz" after you are done modifying it.

Otherwise you will get the following error:

/usr/lib/asterisk/modules/codec_g729a.so: undefined symbol: gzread
Loading module codec_g729a.so failed
Other articles in this category
document I am hearing an echo. What can I do to fix this?
document How do I install the G.729 Codec?
document What is Asterisk?
document I want to execute asterisk command in shell. How do I do that ?
document I want to be able to log the console output. How do I do that ?
document I updated asterisk code from SVN and now I want to go back to older code. How can I do that ?
document ACD Terminology
document Do I need a sound card for music on hold or voicemail?
document How many calls can I get on a system simultaneously?
document Why do I hear echo when I make a call?
document How do I execute Asterisk commands in a shell?
document How do I log console input?
document ACD terminology?
document What is the maximum number of simultaneous calls allowed by Asterisk Business Edition?
document How do I use Windows Messenger with Asterisk?
document Why won't Asterisk start?
document Why am I not getting any audio using SIP?
document Why am I only getting audio in one direction using SIP?
document Why does Asterisk's Playback function not work?
document How can I get additional debug information for SIP?
document Why am I getting dropped calls?
document Why am I getting phantom rings?
document Why am I having DTMF detection problems?
document Why is there a pause after the last DTMF digit?
document Why am I not getting a dialtone using a channel bank and a T1 card?
document Why is my handset not ringing?
document I'm running SELinux and Digium's G729 codec won't load properly. What can I do?
document How can I get additional debug information for IAX?
document How can I get additional debug information for RTP?
document What ports does SIP use?
document What ports does IAX use?
document Does the IAX protocol traverse NAT easily?
document Why am I getting a clicking noise?
document How do I troubleshoot problems over email with Digium's Technical Support staff?
document How do I convert the IVR prompts I've purchased from Digium to a format usable by Asterisk?
document What is fxotune and how do I use it?
document Does Asterisk support faxing?
RSS Powered by KBPublisher (Knowledge base software)