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

[Antonio Modesto]

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. 
> 
> --
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