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...liable 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, headphones combinations. None of them pr...

Could anyone quote a price for the following project. We should be able to generate a specific (say 12Khz) signal at certain intervals (calculated using a price/rate table on a mySQL database) DURING an ongoing conversation. The conversation is to be marked (start and end) with specific signals as well. This is a requirement for special hotel applications where a device counts the signals to

I've tried almost any softphone available on the market with many different PC, soundcard, headphones combinations. None 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

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:

Hi, I'm using *CVS Head version and read the dialplan from 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

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.