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

Antonio Modesto modesto at isimples.com.br
Tue Dec 13 13:07:57 CST 2011


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 <modesto at 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.
>         > > > > > 
>         > > > > > 
>         > > > > > 
>         > > > > > --
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>         > > > > 
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>         
>         
>         
>         
>         
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> 
> 
> 
> 
> 
> 
> -- 
> Atenciosamente
> 
> ____________________
> Roberto Linck
> robertolinck at gmail.com
> (51) 8140-1372
> 
> 
> --
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