On Tue, Oct 12, 2010 at 4:40 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
It is a known-issue that you never know what you're going to get with Asterisk.. yes.
Is there a way to solve this issue without taking away asterisk?
On Tue, Oct 12, 2010 at 8:22 AM, Piotr Sobolewski <sobolewski@gmail.com> wrote:
Hi
When conversation of the call is being saved to a file (record_session) audio tempo in such recording is like slowed down despite that call conversations lacks this issue. Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG If the call comes from IP-Phone, recording is normal. (UA-->FS-->Asterisk) OK
I've tried multiple options for example: <action application="export" data="rtp_rewrite_timestamps=true"/> <action application="set" data="bridge_generate_comfort_noise=true"/>
And setting <param name="rtp-timer-name" value="none"/> in the sofia profiles didn't solved the problem either.
Global Codecs definition.
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU@20i,PCMA@20i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMA@20i,GSM"/>
Is this known issue?
-- Piotr Sobolewski sobolewski@gmail.com
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-- Anthony Minessale II
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-- Piotr Sobolewski sobolewski@gmail.com