Types of bridging

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Earlier I was using asterisk 1.4 and 1.6. In these version it used to
do native bridging and the CPU load was not very high. Now after
switching to asterisk 1.8 it has started to do remote bridging and the
CPU load has often started to peak. Could this be a configuration issue. I have done the same SIP settings
that was earlier there in 1.4 and 1.6. I have 'directmedia=yes' and
'directrtpsetup=yes' in sip.conf and both the peers use the same
codecs and there are no nat issues as well Please help On…

Asterisk Users 3.4 years ago 0 Answers

progressinband, how much extra CPU load?

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Hi everyone, We have an Asterisk 1.4.17 user who has problems with sometimes not getting
a ring tone on the calling phone. We're considering setting progressinband = yes, but would like to know how
much extra CPU load this will require? If anyone can give something even
roughly specific (eg "30% increase") that would be great, rather than just
"lots". Also, are there any ATAs which are known to not work with progressinband =
yes? We have Polycom, Linksys and Audiocode. Thanks for any advice,

Asterisk Users 4.6 years ago 0 Answers

Call recording format

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What format are the actual calls in? Are they in G.711u/a format or
are they in something else (perhaps gsm?) format? I'm asking to find
out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius
wrote:
> Hi All,
> We have a requirement to record over 60 simultaneous calls. Our recording
> facilities are implemented using Monitor() over AMI. The thing we have
> noticed that making 60 simultaneous call recordings using wav CPU load is
> significantly higher (around 2 times…

Asterisk Users 4.8 years ago 6 Answers

Asterisk Playback sound dropping on linphone

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Hi,
I dial on A* from a linphonec to a Playback() extension, then suddenly
the sound stops after a while, without any notice.
I enabled debug both in linphone and A*, and the RTP packets are sent
from A* and received from linphone. It doesn't matter whether I choose
alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. I just need a console scriptable softphone, so maybe there's an
alternative to linphone (which seemed good enough anyway!)...…

Asterisk Users 4.8 years ago 3 Answers