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There is plenty of information on the wiki for setting asterisk up for transferring calls both from the Dail() command, and features.conf. What really seems to be missing, is simply how do you actually perform the transfer? Blind transfers are pretty simple as you only have two obvious steps. How though do you do attended transfers? 1.) You have a call 2.) You dial *2 or whatever you have

Hey, I would like in the course of dial plan logic, to trigger a separate outbound call. If that outbound call is answered, and if that certain key response is detected then it will bridge the incoming call to the newly dialed outbound call. What I want to accomplish is that when a caller dials in, they can enter enter an extension that will call out to a callee's cell phone. When the

I'm really confused on how to use these two options together: A while back: JustRumours edited this page: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe and added a little section about dynamic conferences. the 'e' option is repeated all over the page as the savior of dynamic conferences, maybe I'm just dumb, but can someone tell me if a conference is created with the e

For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available

"In general you can match callerID with the /, but if you don't put anything after the /, then the rule matches "no caller*ID", and if no slash is there at all, it matches "any callerid". " Ok.My question is -> how to match callerid from 001... ? and if don't know how many numbers ? exten => s/0_,Answer don't work- anything else ? tnx Thomas

How do you tell if an ext/SIP account.is in use? For instance can you tell from a Dial() what the status of the line was. I would like to differentiate between in use and unanswered (for someone who is on the phone, but doesn't want to take the call) & ignored/dnd/unanswered but not in use (for someone who isn't at the phone, or just isn't interested in answering the phone.

I made an edit to the wiki: http://www.voip-info.org/wiki/view/Asterisk+tips+campon While I need this solution, and I think that some other people can benefit from various aspects of it, can anyone see if there is a more elegant solution to achieve the same result? Please feel free to edit that example so it is the best for everyone, I'm a true amateur, and I think that example needs a

I'm having a real problem with one of my linksys pap2. On outgoing calls the callee will ring, but caller (pap2) will not here it ring When the callee answers, no audio is transmitted either way. Asterisk reports the call connected and bridged correctly. Now the kicker is that sometimes it works and other times it doesn't. I have had the most luck calling land lines, but sometime

Sorry somehow I wasn't subscribed to the mailing list so I missed any posts, that is fixed now. Anyway I solved my CPU issue problems it turns out that root 3 59.6 0.0 0 0 ? R< 10:16 11:04 [events/0] Before yesterday I had a perfectly (as far as I could tell) functioning PDC with samba, roaming profiles and all. Yesterday I made some changes to the server 1.)

Just starting to enjoy the full features of asterisk, I do have a couple questions though, that I can't seem to find answers for in the wiki, just wondering if someone could light my way. after a caller has made their choice of options in the dial plan, I would like them to be placed on "hold" (music, not ringing) while the system processes through the rest of the dial plan

Don't know how that last email took off, but that wasn't suppose to happen. Any the root 3 59.6 0.0 0 0 ? R< 10:16 11:04 [events/0] is a kernel thread that handles weird irq stuff. I just disabled onboard sound and USB and it went away. I'm still left with my other problems though, here they are 2.) I'm having LMB and DMB problems on the server,