Hi Antonio, I'd never had used extensions.ael but in extensions.conf, using Macro I always set '__TRANSFER_CONTEXT' to the same context of exten and it works well.
2011/12/13 Antonio Modesto <[email protected]> > ** > Hello everybody, > > I found that if i write my macro in the extensions.conf (not in ael), > the atxfer works well, the problem is that ael uses gosub instead of the > Macro() application, which doesn't change the current context. Does anybody > know if i can do anything to solve this? I know if i rewrite all my macros > in the common way, it will work, but that's a lot of coding for me. > > > > On Mon, 2011-12-12 at 08:57 -0200, Antonio Modesto wrote: > > Nothing? > > > On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote: > > > > > Hi There, > > I'm still having this problem, Does somebody know what can be > happening? > > > Regards. > > On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote: > > Hello, > > The exten is the parameter passed to the macro, which contains the sip > device name. I'll change the name to another less confusing. > > * Alexandre, também sou brasileiro hehe, notei que você já escreveu um > livro sobre asterisk, será que você poderia me ajudar com esse problema? Já > tem alguns dias que estou na luta aqui hehe. > > On Fri, 2011-11-11 at 08:45 -0200, Alexandre Keller wrote: > > You're using ${exten} inside your macro, you should use ${EXTEN}. > -- > Atenciosamente, > > ALEXANDRE KELLER > > > http://twitter.com/alexandrekeller > http://www.facebook.com/alexandre.keller.BR > > "Dinheiro é a consequência de um trabalho bem feito e não o motivo para se > fazer um bom trabalho." > > > *P Antes de imprimir pense em seu compromisso com o Meio Ambiente.* > > On 11/11/2011, at 08:38, Antonio Modesto wrote: > > On Mon, 2011-11-07 at 09:12 -0600, Danny Nicholas wrote: > > It can have to do with either the telephones dial plan or the context in > the Asterisk dial plan combined with your features.conf settings. > > > I noticed that my problem occurs when i use a macro to dial sip devices, > my dialplan is like this: > > - Each sip device has its own context > - This context includes the outgoing call contexts that this extension can > use for making calls and includes a context called "ramais", which has the > dial plan to call another extensions, it uses a macro to do this. > > Here is the configuration for my extension "modesto" : > > # sip.conf > [modesto](default_extension) > username=modesto > context=modesto > callerid="modesto" <106> > callgroup=4 > pickupgroup=4 > > # Default extension template > type=friend > dtmfmode=auto > host=dynamic > disallow=all > allow=ulaw > allow=alaw > deny=0.0.0.0/0.0.0.0 > permit=192.168.1.0/255.255.255.0 > canreinvite=yes > qualify=no > callcounter=yes > > > # context for SIP/modesto > context modesto { > includes { > vivo; > tim; > oi; > claro; > vivoddd; > timddd; > oiddd; > claroddd; > embratel; > embratel2; > }; > includes { > ramais; > }; > }; > > # Although the problem is occurring also for others contexts included, > i'll show only the "ramais" context, which is used to call local extensions: > > context ramais { > 101 => &dial_sip(suporte1); > 102 => &dial_sip(suporte2); > 103 => &dial_sip(suporte3); > 105 => &dial_sip(suporte05); > 106 => &dial_sip(modesto); > 107 => &dial_sip(gustavo); > 108 => &dial_sip(pauloh); > 109 => &dial_sip(fernanda); > 111 => &dial_sip(marcos); > 112 => &dial_sip(thiago); > 115 => &dial_sip(helder); > 116 => &dial_sip(atendimento01); > 117 => &dial_sip(atendimento03); > 118 => &dial_sip(atendimento02); > 119 => &dial_sip(marlon); > 120 => &dial_sip(suporteemp); > 122 => &dial_sip(telemais); > 123 => &dial_sip(casagustavo); > 127 => &dial_sip(manutencao); > 128 => &dial_sip(guilherme); > 129 => &dial_sip(marcelo); > 130 => &dial_sip(rafael); > 132 => &dial_sip(netita2); > 133 => &dial_sip(unotel); > > }; > > If I use the Dial() application instead of this macro, it works well. I > noticed that when I use the macro and try to transfer a call (The problem > occurs only for the calling party, the called party can do transfers with > no problems), asterisk tries to find the extension in the <macro-name> > context and of course, there is no dialplan to call the extensions there. > > > Here is the dial_sip macro: > > 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; > }; > > Thanks in advance. > > > > -- > _____________________________________________________________________ > -- 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 > > > -- > _____________________________________________________________________ > -- 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 > -- Atenciosamente ____________________ Roberto Linck [email protected] (51) 8140-1372
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