I’ve got Asterisk PBX system consisted of 2 SIP clients (X-Lite softphones running on Win XP). Calling and voice transmission in calls works OK, but I got problems in playing .gsm sample sound files in the call.
I’ve set up auto answer with playback of some .gsm sound file on one extension and when I call that extension, it automatically answers but plays that sound file extremely bad, with mumbling instead of real sound. When I change the played sound file, I just got another, different mumbling. I’ve realized that function of playing files goes OK, but only with the extremely low quality, the voice recorded in the file is unrecognizable.
I am facing the same issue with AsteriskNOW 1.0.2 with GUI2.0.
The PC I am using is a 800MHz AMD with 384MB RAM.
Analyzing the output of “show modules” CLI command it seems that the issue is related to the format_gsm.so. Only when this format is active the sound is bad. Playing a .WAV (the default for music on hold) works great.
Calls with both GSM and G.711 (alaw) also works great.
I checked the load on the CPU (~5%) and the free RAM is 100MHz.
I also suspect the slow hard disk access
Playing a .wav file in my setup also makes a problem. I realize that this all issue is related to something about codecs, but I don’t know what. It seems that my Asterisk doesn’t play sound files well I didn’t make any changes to other sample files besides extensions.conf and sip.conf .
I have read that some of this issue is from compiling gcc
If you search the forum for gsm and gcc, you may find an answer on it
sorry I just didnt have time to do the search for you
Try first to record your files in wav format and then you have to use a tool called SOX to convert your files into GSM using this command:
sox foo.wav -r 8000 -c 1 foo.gsm after downloading it.
you can download it from here: sourceforge.net/project/showfile … p_id=10706
[/url]
I didn’t make any changes to other sample files besides extensions.conf and sip.conf .