John Faubion wrote: >> We do have full features on our lines so both lines are free once the >> transfer is complete. We also have toll calls on our lines so it would >> not be a problem, so I do not have to worry about AT&T dropping the >> > > The issue really isn't whether you have the ability to make toll calls on > your line. The concern here is in what the regulatory agencies call "toll > bridging" which is using a system to relay a call from one local calling are > to another local calling area to avoid a toll charge. This is one of those > gray areas that can become a problem if your not careful. The problem comes > up if you have customers that can call you as a local call and you are > forwarding them on to another party that is a local call for you but would > be a toll call for the customer. This is essentially what toll bridging is > about. Now your not likely to have to worry about the legal ramifications of > this since your merely connecting the customer with an extension of your > company, namely your salesman. Where this could become a problem for you > would be in transferring the customer using the same pots line. The reason > is that AT&T is handling the transfer. When you transfer the call, it > essentially becomes a new call. The main difference is that you have > provided the called number. So the software in the Class 5 (End office) > switch, takes the number you provide and runs the call through its routing > translations (similar to the Asterisk dialing plan) and if it determines > that the destination number is outside the originators Local Area Transport > Area or LATA, then it will either drop the originator to a message that > says, "You must first dial a 0 or 1 before calling this number" or it may > deny the transfer allowing you to stay connected to the customer. Neither > one looks very professional. The only way around this would be to provide > another line or trunk to pass the call down. Now if your not in an > overlapping LATA this probably isn't an issue. > > John you a right about the LATA I know I am in one LATA 536 or 538 for > eastern OK. But I do not know the LATA on the Wireless which is now AT&T. So > I will keep a watch out for it. Thanks for the tip! > >> The only way I can get it to work is by have the call on the 1st >> line then transfer it out on the 2nd line. After that is complete both >> lines are free. >> > > Are you saying that you are able to route a call from line 1 to line 2 and > have the call transfer, thus freeing the lines or that once the call > completes the lines are freed? I've never seen the first scenario. The > second scenario is the normal behavior. > > I am saying here that I can transfer the call from line 1 to line 2 and once > I transfer off the asterisk box it frees the two phone lines. My whole > arguement was to find a solution for doing this automatically on the basises > of dial an extension which can just transfer it to the cell phone. So ext > 4001 cell-1 ext 4002 cell-2 etc. I do not mid doing it manually. But thanks > for the help! > >> Can you give an example of creating an extension which points to a cell >> phone. Secondly how can you have if no one answers an extension it dials >> the cell number next. That maybe answered in the example. >> > > In extensions.conf use something like this. > [global] > SIP-PROV = "sip.urprovider.com" > ; Now set the call forward numbers > CFN21 => "5555551234" ; These are normally set in an external file > > [internal] > exten => 21,1,Macro(stdext,${SIP/21},${CFN${EXTEN}}) > > [macro-stdext]; > ; Standard extension macro: > ; ${ARG1} - Device(s) to ring > ; ${ARG2} - Our call forward number > exten => s,1,Dial(${ARG1},10) > exten => s,2,Goto(s-${DIALSTATUS},1) > exten => s-NOANSWER,1,GotoIf($[${LEN(${ARG2})}>0]?s-CFWD,1) > exten => s-NOANSWER,2,Voicemail(${MACRO_EXTEN},u) > exten => s-BUSY,1,Voicemail(${MACRO_EXTEN},b) > exten => s-CFWD,1,Dial(SIP/[EMAIL PROTECTED],20) > exten => s-CFWD,2,Goto(s-NOANSWER,2) > exten => _s-.,1,Goto(s-NOANSWER,2) > exten => a,1,VoicemailMain(${MACRO_EXTEN}) > > > There is more to this but this should show the basics of what we use. I > store my Call Forward Numbers (CFN) in an external file. This allow me to > update the file externally (currently with a web interface but as soon as I > get the prompts recorded it will be done with an IVR) and then just reload > the extensions to activate the new numbers. Also I using SIP for pretty much > everything. Our TDM400 doesn't even have modules, it's just there for > timing. However you should be able to convert the SIP calls to ZAP calls for > you use. The internal context is included in our default context. Dialing > extension 21 calls the stdext macro. This dials the local extension first. > If not answered after 10 seconds, we check to make sure we have a phone > number to send the call out with. If not we send it on to voice mail. > Otherwise we send it to the s-CFWD. The check listed here is a very > rudimentary check but again I hope you get the idea. Next we try the call to > the CFN. If not answered in 20 seconds, then we send it to voice mail. > Finally if the user presses the star button during the attempt, we send them > on to Voicemail mail so they can check their messages. > > Hopefully this helps. > > John > >
Yeah this is similar to what I was looking for I will have to play with it this weekend. Thank bro! > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users