Steve Totaro wrote: > On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton < > dfullertaster...@shorelinecontainer.com> wrote: > >> Steve Totaro wrote: >>> On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton < >>> dfullertaster...@shorelinecontainer.com> wrote: >>> >>>> Tzafrir Cohen wrote: >>>>> On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote: >>>>> >>>>> You have a small typo: >>>>> >>>>>> exten => _.,1,Dial(Zap,g1,${EXTEN}) >>>>>> exten => _.,2,Dial(SIP,Provider,${EXTEN}) >>>>> exten => _.,1,Dial(Zap/g1/${EXTEN}) >>>>> exten => _.,2,Dial(SIP/Provider/${EXTEN}) >>>>> >>>>> ('/' instead of ',') >>>>> >>>> While this will work, be aware that there are circumstances where you >>>> may end up calling the number twice, once through each provider. One >>>> example is if the number you dial is busy, that progress will be passed >>>> via the PRI to asterisk and the dialplan will continue to the next >>>> priority. In this case, dialing the number again through the SIP >>>> provider. To avoid this you will need to use some dialplan logic and >>>> check the result of the DIALSTATUS variable. See this page for examples: >>>> >>>> http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS >>>> >>>> -Dave >>>> >>>> >>> Good point. >>> >>> I was unaware that "busy" back from a TDM circuit would progress in the >>> dialplan rather than going to the h exten. >>> >>> What other cases are there like that? >> It is my understanding (through trial and error, reading, etc) that any >> Dial command that does not result in an answered state will continue in >> the dialplan after a timeout (if specified) or some sort of progress is >> received. If the called channel results in an answer then dialplan >> processing stops as soon as one party hangs up (unless the g option is >> specified). >> >> This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI >> PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon >> as the dial is complete so you won't be able to use this trick under >> normal circumstances. >> >> -Dave >> >> > True I guess except that if the call fails as the OP posted, because the PRI > is down, it should work then right?
I believe so, I haven't tried it. I imagine DIALSTATUS would be either CHANUNAVAIL or CONGESTION. > > Another thing. For outbound calls, I do not have a timeout. So the user > hangs up when they are ready, or when the other side hangs up or gets > congestion, which amounts to the h exten, or am I not correct. I can't answer to the use of the h exten, I've never used it. > Why have a timeout on outbound dialing (unless you are a dialer app?) It is > not like voicemail where you want it to ring for so many seconds and then > roll to VM. You usually wouldn't use a timeout for outbound PSTN calls. I only mentioned it to try to be as complete as possible. -Dave _______________________________________________ -- 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