I have found that you get good results by setting a per-device GROUP_COUNT(), which prevents dialling if it is non-zero, and setting call-limit to 999.
In Asterisk 1.0.x there were separate in- and out-bound call limits, but IIRC this was pretty broken, and was removed. See http://www.voip-info.org/wiki/view/Asterisk+func+group Hope that helps. Steve 2009/3/31 carl Lougher <[email protected]>: > > Yeah but doesnt help for extensions that have or require call-limit=1. > > --- On Tue, 31/3/09, carl Lougher <[email protected]> wrote: > >> From: carl Lougher <[email protected]> >> Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >> <[email protected]> >> Date: Tuesday, 31 March, 2009, 2:20 AM >> >> We use call-limit set to 1 for hints. I guess i'll look >> into the dtmf method and debug the linksys phone to see what >> it uses for attended transfers. >> >> Cheers!!!! >> >> --- On Mon, 30/3/09, Mark Michelson <[email protected]> >> wrote: >> >> > From: Mark Michelson <[email protected]> >> > Subject: Re: [asterisk-users] Call-limit=1 breaks >> attended transfer >> > To: "Asterisk Users Mailing List - Non-Commercial >> Discussion" <[email protected]> >> > Date: Monday, 30 March, 2009, 10:50 PM >> > carl Lougher wrote: >> > > Howdy, >> > > Was there ever a fix for this? >> > > >> > > I have Trix 2.6 running asterisk 1.4 and have to >> set >> > an extension with call-limit=1. However that user can >> no >> > longer do attended transfers from Linkys 962 ip >> phone. >> > > >> > > Is there anyway around this? >> > > >> > > Cheers, >> > > Taff.. >> > > >> > >> > Yes, set call-limit to something else :P >> > >> > Seriously though, there's no "fix" for that since it >> is >> > behaving exactly as it >> > should. When attempting to transfer the call, Asterisk >> has >> > no way of knowing >> > that the new SIP INVITE it receives (in order to call >> the >> > transfer target) is an >> > attempt to transfer the call. It appears that the same >> SIP >> > peer is attempting to >> > make a second call. Since the call-limit is set to 1, >> > Asterisk rejects the >> > second call attempt. >> > >> > I haven't tried this yet, but it may actually be >> possible >> > to use DTMF transfers >> > when the call limit is that low since Asterisk is the >> one >> > that actually >> > initiates the new call to the transfer target instead >> of >> > the transferer's phone. >> > To use DTMF transfers, you need to set a DTMF sequence >> in >> > features.conf and use >> > the 't' or 'T' flag (depending on which party should >> have >> > the ability to >> > transfer the call) in your calls to Dial() or >> Queue(). >> > >> > Why do you have the call-limit set to 1, anyway? >> > >> > Mark Michelson >> > _______________________________________________ -- 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
