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

[Danny Nicholas]

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.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ramiro Paz
Sent: Monday, November 07, 2011 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with Atxfer for the calling party

 

Hi:

Same problem here (asterisk doesn't wait all the digits be typed when making
a call transfer). Does anybody knows something about this? Thanks in
advance.

Greetings,

Ramiro PAZ
MASTERLINE LOGISTICS



2011/11/1 Antonio Modesto <modesto at isimples.com.br>

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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