[Asterisk-Users] Supervised transfer problem.

Cesar Garcia cesar.garcia at idecnet.com
Tue May 3 05:21:02 MST 2005


Hi All.

Thanks¡¡¡, Now i can blind transfer calls with "#" key, but in 
featuresmap said

 >>blindxfer => #1                ; Blind transfer
 >>disconnect => *0               ; Disconnect
 >>automon => *1                  ; One Touch Record
 >>atxfer => *2                   ; Attended transfer

but any key with * during call, (budgetone or x-ten)

produce this in console

Attempting native bridge of SIP/u0002-5fdd and SIP/u0001-eb16

And only "#" key its ok to transfer (not #1 how is stablished in 
featuresmap).

any more configuration in asterisk? i dont know what more to do.


César García.
    Director de Sistemas, IdecNet S.A.
    Centro de Gestión de Red.
    Edificio IdecNet. C/Juan XXIII 44.
    E-35004, Las Palmas de Gran Canaria,
    Islas Canarias - España.
    Tfn:  +34 828 111 000 Ext: 340

Arunachala escribió:
> Hi, 
> 
>    Please include "tT" options in your Dial statements in extensions.conf.
> 
> Example:
> 
> extensions.conf
> 
> [default]
>  
> exten => 0000,1,Dial(SIP/u0001&SIP/u0004,20,tT)
> exten => _0XXX,1, Dial(SIP/u${EXTEN},20,tT)
> exten => 828112070,1,Dial(SIP/u0001,20,tT)
> exten => 828112071,1,Dial(SIP/u0004,20,tT)
> 
> These indicate to asterisk that caller & the callee are both allowed
> to transfer the call.
> 
> Regards,
> Arun
> 
> On 4/27/05, Cesar Garcia <cesar.garcia at idecnet.com> wrote:
> 
>>Hi all.
>>
>>I am new in the list and i believe i have read enough to run an asterisk
>>pbx good, but i have a problem.
>>
>>My instalation is enterely SIP based and i am trying now with
>>grandstream budge tone 102 because with x-lite softphone i cannot get
>>transfer, supervised or not, be fine.
>>
>>Few question:
>>
>>Is supervised transfer supported by SIP channel in 1.0.7 stable release?
>>
>>Why i cannot obtain results with the "hot keys" listed in featuresmap?.
>>[featuremap]
>>blindxfer => #1                ; Blind transfer
>>disconnect => *0               ; Disconnect
>>automon => *1                  ; One Touch Record
>>atxfer => *2                   ; Attended transfer
>>
>>i dont obtain results with this hotkeys, but pickup key *8 is ok.
>>
>>dtmf is inband
>>
>>Thanks to all in advance and for this great work¡¡¡
>>
>>this is my sip.conf and extensions.conf
>>
>>sip.conf
>>
>>[general]
>>port=5060
>>bindaddr=0.0.0.0
>>context=default
>>srvlookup=yes
>>dtmfmode=inband
>>disallow=all
>>allow=all
>>language=es
>>
>>[u0001]
>>type=friend
>>username=u0001
>>secret=xxxxxx
>>auth=md5
>>callerid="Cesar Garcia" <0001>
>>host=dynamic
>>callgroup=1
>>pickupgroup=1
>>nat=yes
>>canreinvite=no
>>
>>------------------
>>
>>extensions.conf
>>
>>[default]
>>
>>exten => 0000,1,Dial(SIP/u0001&SIP/u0004,20)
>>exten => _0XXX,1, Dial(SIP/u${EXTEN},20)
>>exten => 828112070,1,Dial(SIP/u0001,20)
>>exten => 828112071,1,Dial(SIP/u0004,20)
>>
>>--
>>
>>César García.
>>    Director de Sistemas, IdecNet S.A.
>>    Centro de Gestión de Red.
>>    Edificio IdecNet. C/Juan XXIII 44.
>>    E-35004, Las Palmas de Gran Canaria,
>>    Islas Canarias - España.
>>    Tfn:  +34 828 111 000 Ext: 340
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> 
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