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I am developing a Call Center with AMI, I have already redirect and create a call, but I want to implement a functionality so I can supervise the call (spy). I have been studying the function chanspy of asterisk. Does any one know how to execute the function Chanspy(SIPXXX) throw AMI? Best Regards Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070616/eaf5a7d5/attachment.htm

...led sip debuggint (fully below) and notice that I get: SIP/2.0 488 Not Acceptable Here Warning: 399 SDP Not Acceptable Lots of info about the 488 implies some codec issue between the endpoints, so I changed my sip.conf [general] to only permit ulaw, as well as the same in the phone's config (SIPxxx.conf). Didn't help. Since it's cleaner this way, this is how I currently have left it. Strangely, transfers work if they come from a ZAP channel TO a queue or directly to voicemail (via an extension) but will NOT work if anything is being sent directly TO a SIP client. I also tested with...