[Asterisk-Users] voicepulse "silence" during conversations

[Race Vanderdecken]

Yes, Ironic isn't it. That SIP/VOIP is so digitally clear that you don't
hear the pops, cracks and whistles of the old analog phones. The only
analog is from the human to the machine. The old analog phone humans
hear it, soon there will another generation of humans who have never
used an analog phone. 

Anyone remember the transition from long distance operators to direct
dial. Or from pulse to touch tone? Back in 1992 I tried to make a
calling card call using a rotary phone in Alabama, where they had 5
digit dialing. I was stumped looking at a phone with no pound/# sign on
it.

I first noticed this silence quirk when I was working with a 3COM SIP
phone back in 2000. The crystal clear voice and silence made me feel
like the phone was not working or that the other person had hung-up.

You also have to be careful of background noise in the room; phones with
good microphones will let the other end here everything going in the
room you are in.

Race "The Tyrant" Vanderdecken

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sean
Kennedy
Sent: Wednesday, March 09, 2005 4:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] voicepulse "silence" during conversations

Hi all, I'm running Asterisk 1.0.0.  I am a customer ( and supporter ) 
of voicepulse.  For me, it works perfectly, but one of my customers 
noticed a small problem:  During a conversation, when the otherside 
isn't talking, it's almost like the mic turns off. 

Not that big of a deal I know, and the more I think about it, the more 
this seems a voicepulse issue.   But in the off chance this could be 
something on my end:

Asterisk 1.0.0
Connecting to voicepulse via iax, ulaw codec
Polycom 500 IP SIP phone, ulaw codec

I'll be honest, I don't notice it at all, but my customer does, and I'd 
like to make them as happy as I can with this system. 

Also ( I would feel silly making another thread out of this ) what are 
the common reasons for echo between sip phones going through two 
different asterisk servers?  As in phone -> asterisk A -> asterisk B -> 
phone.  I've been searching for it, but I'm not having much luck.

Thank you, any help is greatly apprecaited!

Sean
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