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Hi, Companies that offer telephone service to the public are obliged to offer tapping to all kind of authorities. Does anyone know how to tap in Asterisk? I.e. record (or copy) a conversation based upon their telephone number? Thanks Dan _________________________________________________________________ Stay in touch with absent friends - get MSN Messenger http://www.msn.co.uk/messenger

I used to be able to pass dial strings to IAX2 providers with # characters at the end of the string. This is how we end dial strings for international calls. So, I would like to be able to selectivity chop off any # characters at the end of string, only if they exist. Basically as follows (chopping off the leading '9' with ${EXTEN:1} syntax: EXTEN from Phone EXTEN for Dial String

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

I have a client who needs an application for there field techs to call in when they arrive on site and when they leave. The logic behind it seems pretty simple. I am going to write something in AGI to capture some DTMF tones and update this data into MySQL to run some reports from. But here's my initial problem. I have started to create a simple AGI script to capture the CallerID, but I

Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of "power hubs", but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be

Hi, Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs running g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa