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...ed (start and end) with specific signals as well. This is a requirement for special hotel applications where a device counts the signals to calculate a price for the ACTUAL (after successful connect) conversation. We believe, this task may require some source code modifications on Asterisk. Thanks, Cenk Yabas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050606/e372f710/attachment.htm

...ond most cost-effective and reliable solution was Sipura SIP Analog adapters, partially because they use cheap analog phones and you can hide them under a desk where they will not get ruined when an agent spills their half gallon of Mountain Dew all over. MATT--- -----Original Message----- From: Cenk Yabas [mailto:cenk.yabas@globalyayincilik.org] Sent: Saturday, June 11, 2005 10:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice quality of Softphones vs. IP Phones and Gateways. I've tried almost any softphone available on the market with many different PC, soundcard...

Can anybody please answer this. Both clients are behind different NAT's. One of them starts a SIP call to the other through Asterisk. Asterisk sets up the call. Issues reinvite and connects them together. After this point does the media stream flow through Asterisk or Peer to Peer? Does such a call use any system resources of Asterisk server after connection? Thank you in advance.

...of them prooved production reliable in a call center environment. I've also tested many IP Phones and Gateways. Even the cheapest one supplies much better quality. Is this a fact or am I missing a point. I would certanly prefer a softphone because of cost and simplicity in CTI applications. Cenk. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050611/17dc354c/attachment.htm

In the dial application when configuring the Limit parameter: L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) I want to read 'z' from database, based on the dialed number. How is this possible? Cenk. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050611/32bdc5cb/attachment.htm

...rom MySQL. I'm making A-Z termination to over 4000 different country and city codes. I have 3 different dialing rules depending on the price level of the dialed number. Should my extensions table contain 4000 lines? Is this realistic? Or is there any other (more clever) way doing this? Regards, Cenk. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050613/4b8ac339/attachment.htm

Hello everyone, I've look at the latest nightly snapshot but I could find this bug is fixed on the source code. Is there any progression to fix that issue. This bug is open for almost 10 years, come on guys :) >List: openssh-unix-dev >Subject: Re: killing remote commands >From: HAUTREUX Matthieu <matthieu.hautreux () cea ! fr> >Date: 2010-10-21