Hi: Same problem here (asterisk doesn't wait all the digits be typed when making a call transfer). Does anybody knows something about this? Thanks in advance.
Greetings, *Ramiro PAZ MASTERLINE LOGISTICS *** 2011/11/1 Antonio Modesto <[email protected]> > ** > Good morning, > > I have not solved this problem yet, but, I found that the source of > the problem are my macros. For example, I have this context: > > context ramais { > 101 => &dial_sip(exten1); > 102 => &dial_sip(exten2); > 103 => &dial_sip(exten3); > }; > > All these extensions use the dial_sip macro, I have changed this context > to use the Dial application instead of dial_sip macro, it worked fine. The > problem is that when i use the macro, the current context is changed to the > dial_sip context, the dial_sip context is automatically created by asterisk > when i use any macro and of fact this context doesn't have the ramais > context included. Is there some way to specify on which context the macro > will run? > > > On Mon, 2011-10-31 at 09:09 -0200, Antonio Modesto wrote: > > Good Morning, > > I have an asterisk18-1.8.7.1 running on a FreeBSD 8.2-STABLE, and it is > working well so far, i'm just having some problems with atxfer. > > I have written this macro to dial sip extensions: > > macro dial_sip(exten) { > Verbose(2,"==> Chamando a MACRO dial_sip - ponto 1 macros.ael > <=="); > Verbose(4,"====> Macro dial_sip iniciada."); > ChanIsAvail(SIP/${exten}); > Verbose(2,"==> ${AVAILORIGCHAN}"); > > if ("${AVAILORIGCHAN}" != "") > { > Verbose(4,"====> SIP/${exten} parece estar disponivel, vou > disca-lo agora."); > Set(FromExt=${CALLERID(num)}); > System(/bin/sh /var/spool/asterisk/calllog/log.sh > SIP/${FromExt} SIP/${exten} SIP-TO-SIP); > Verbose(4,"====> System status: ${SYSTEMSTATUS}"); > Dial(SIP/${exten},${SIP_DIAL_TIMEOUT},Ttr); > Hangup(); > } > else > { > Verbose(2,"====> SIP/${exten} nao esta disponivel."); > Hangup(); > }; > > > NoOp("From ${MACRO_EXTEN} to ${exten}); > System(${CALLLOGDIR}/log.sh ${exten}); > > return; > }; > > It is working, but the calling party is not able to transfer the calls > because asterisk doesn't wait all the digits be typed, it tries to transfer > the call when the first digit is pressed (We use 3 digits extensions): > > [Oct 31 09:04:01] WARNING[2926]: features.c:2315 builtin_atxfer: Extension > '1' does not exist in context 'dial_sip' > == Spawn extension (dial_sip, ~~s~~, 11) exited non-zero on > 'SIP/modesto-0000000d' > [Oct 31 09:04:03] WARNING[2926]: features.c:2319 builtin_atxfer: No digits > dialed for atxfer. > > Does anyone have suggestions? > > Regards. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
