Hi there,
i called one asterisk server from another asterisk server. The calling
server played back a audio data und the answering server recorded the audio
sample using record() function.
I tried both ISDN, VoIP connections. Camparing with the original audio data,
the recorded samples from both
...@lists.digium.com] *On Behalf Of *Felix Dong
*Sent:* Tuesday, March 15, 2011 4:19 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] signal amplified by asterisk
Hi there,
i called one asterisk server from another asterisk server. The calling
server played back
is”.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
*Sent:* Tuesday, March 15, 2011 4:29 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users
] *On Behalf Of *Felix Dong
*Sent:* Tuesday, March 15, 2011 4:19 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] signal amplified by asterisk
Hi there,
i called one asterisk server from another asterisk server. The calling
server played back a audio
.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
*Sent:* Friday, March 04, 2011 8:55 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Loudness of recorded wav-audio
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
*Sent:* Monday, March 07, 2011 6:07 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Loudness of recorded wav-audio
I tried to ajust the tx- and rxgain for the sip peer
Hello,
I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it
in wav-audio at the Asterisk server. I found the loudness level of the
recorded audio was too high comparing with the orginal audio. How can I
ajust it, so that there will be no amplifier used for recording.
Thanks
Thank you! How can I reduce the RXgain?
Am 04.03.2011 um 15:21 schrieb Danny Nicholas da...@debsinc.com:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 2:31 AM
To: asterisk-users
the incoming volume by 4 decibels. You’ll have to do a “sip reload”
for this to take effect.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 8:33 AM
To: Asterisk Users Mailing List - Non
Hello,
can anyboby tell me, how can I disable the echo cancellation for sip?
thx a lot...
best regards,
Felix
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Hi guys,
the function Echo() did work on CAPI, but doesn't work for SIP connection.
Can anybody help?
thanks a lot.
best regards,
Felix
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New to Asterisk?
...@lists.digium.com] *On Behalf Of *Felix Dong
*Sent:* Wednesday, February 16, 2011 4:48 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] function Echo() doesn't work
Hi guys,
the function Echo() did work on CAPI, but doesn't work for SIP connection.
Can anybody help?
thanks
.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
*Sent:* Wednesday, February 16, 2011 5:14 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] function Echo() doesn't work
...@lists.digium.com] *On Behalf Of *Felix Dong
*Sent:* Wednesday, February 16, 2011 5:33 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] function Echo() doesn't work
* == Using SIP RTP CoS mark 5*
*-- Executing [1174614@von-voip-provider:1
...@lists.digium.com] *On Behalf Of *Felix Dong
*Sent:* Wednesday, February 16, 2011 5:33 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] function Echo() doesn't work
* == Using SIP RTP CoS mark 5*
*-- Executing [1174614@von-voip-provider:1
[mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
*Sent:* Wednesday, February 16, 2011 6:22 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] function Echo() doesn't work
I tried to set allow=all in sip.conf
Hello,
could I adjust the Rx and Tx gains for SIP and CAPI? If it is possible, how
should I do it?
Thanks a lot.
best regards,
Felix
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New to Asterisk?
how did you increase it?
Am 16.02.2011 um 00:11 schrieb Hans Witvliet h...@a-domani.nl:
On Tue, 2011-02-15 at 18:06 +0100, Felix Dong wrote:
Hello,
could I adjust the Rx and Tx gains for SIP and CAPI? If it is
possible, how should I do it?
Thanks a lot.
best regards,
Felix
Hallo everybody,
I got a question to asterisk 1.6. Is it possible to playback a Audiofile in
uplink and to record the downlink channel in another Audifile at the same
time?
If it is possible, how should I do it? Please explain it.
Thank you for your help to my thesis!
best regards,
Felix
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How can I use the application Monitor() in the Python AGI skripts?
Thanks a lot.
best regards,
Feilx
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