As explained in the posts before, this tread was solved.
Thanks. On Tue, 2011-12-13 at 17:07 -0200, Antonio Modesto wrote: > On Tue, 2011-12-13 at 16:35 -0200, Roberto Linck wrote: > > > 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. > > > Thanks, it worked, the MACRO_CONTEXT variable was empty, I've set the > value to my extensions context and it worked fine. > > > Thanks. > > > > > 2011/12/13 Antonio Modesto <mode...@isimples.com.br> > > > > 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 > > robertoli...@gmail.com > > (51) 8140-1372 > > > > > > -- > > _____________________________________________________________________ > > -- 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? 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