the issue is lack of sidetone u can google sidetone
sidetone is feedback u get from the mike to your earpiece that the fone generates to let u know the circuit did not go dead when people stop talking
i find the lace of sidetone extremely annoying and so will many customers
with asterisk i have found lack of sidetone on the grandstream budgetone i have found perfect sidetone on the cisco 79xx and also on the sipuras
Race Vanderdecken wrote:
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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-- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE ---- I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
