asterisk users - Jan 2005 - G.729? Worth it? -- YES --

im using  g729, but the bw usage is ~26 kbps per call, my gateways (cisco)
support g723 and the bw between the gateways is ~18 kbps per call. Much
better than the ~62 kbps of the g711. if you plan to be a voip provider you
"must" go with compression codecs, especially if you want your
customers to
browse the  internet while having a call.
i.e. : We give voip phones (grandstream) to our VIP internet customers (512k
- 1Mb bw), and create a tunnel (QoS) for voip traffic of 128 kbps, with
compression codecs the  customer can have 4 - 5 simultaneous calls, without
compression codecs 2.
BTW, i want to use only g723 but coudnt find licences for it, where can buy
it in groups of 10 - 20 instead of 1000's ?
Miguel 


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of denon
Sent: Wednesday, January 19, 2005 2:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] G.729? Worth it?


At 01:49 PM 1/19/2005, you wrote:
>There are systems that use G.711 when traffic is light, but
>switch to compression codecs under heavy traffic to conserve
>bandwidth.  I don't know how/if this can be done in Asterisk.
>
>--Stewart

I don't think there's anything like that built into * as it is now, but
it
would be pretty trivial to write a script to handle such a thing. Effective 
monitoring of the traffic  conditions may require a bit more work .. 
monitoring latencies, and maybe monitoring current utilizations via snmp on 
your core routers/switches/etc. A quick probe to the management interface 
would also give you some insight on current call volume, of course.

-d

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Hi.

I have the following weird phenomenon on a standard * @ home installation.

I use X-PRO on 2 or 3 computers.


Blind Transfers:
With incoming calls: I click transfer then dial extension the click transfer
again.
Result: Hangs up on incoming caller.

With outgoing calls: I click transfer then dial extension the click transfer
again.
Result: Transfer is ok, dialed extension connects to outgoing call.

Assisted Transfers:
Place incoming caller on hold. 
Get new line and dial extension.
Chat with extension then click transfer and the line number of incoming
caller.
Result: Incoming caller can hear new extension ok, but new extension can
only hear music.

OR
Place incoming caller on hold. 
Get new line and dial extension.
Chat with extension then place him on hold too.
Return to incoming caller, click transfer and the line number of dialed
extension.
Result: Incoming caller can hear music only, but new extension can incoming
caller.

This is very perplexing. It is like the XPro is interacting with * in a way
that it is transferring the "MOH" channel, not the person. And the
order is
weird. Transferring 1 to 2 gives reverse result to transferring 2 to 1.

When I had individual SIP accounts to our VoIP provider's * server, rather
than our own, everything worked fine.

Please help if you can, this is baking my noodle!!!




Mike Sander
Operations Manager

Suite 4 / 38-48 Waterloo St
Surry Hills N.S.W 2010
Phone:(02) 8307 8877
Fax:(02)93182254
Mobile:0401 010 289
Email: mike@corporatebankinginternational.com
Website: www.corporatebankinginternational.com

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