[asterisk-users] Problem with Atxfer for the calling party

Antonio Modesto modesto at isimples.com.br
Tue Dec 13 11:57:08 CST 2011


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.
> > > > > 
> > > > > 
> > > > > 
> > > > > --
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