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${EXTEN},20)
exten => 828112070,1,Dial(SIP/u0001,20)
exten => 828112071,1,Dial(SIP/u0004,20)
--
C?sar Garc?a.
Director de Sistemas, IdecNet S.A.
Centro de Gesti?n de Red.
Edificio IdecNet. C/Juan XXIII 44.
E-35004, Las Palmas de Gran Canaria,
Islas Canarias - Espa?a.
Tfn: +34 828 111 000 Ext: 340
Hi,
Please include "tT" options in your Dial statements in
extensions.conf.
Example:
extensions.conf
[default]
exten => 0000,1,Dial(SIP/u0001&SIP/u0004,20,tT)
exten => _0XXX,1, Dial(SIP/u${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@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${EXTEN},20)
> exten => 828112070,1,Dial(SIP/u0001,20)
> exten => 828112071,1,Dial(SIP/u0004,20)
>
> --
>
> C?sar Garc?a.
> Director de Sistemas, IdecNet S.A.
> Centro de Gesti?n de Red.
> Edificio IdecNet. C/Juan XXIII 44.
> E-35004, Las Palmas de Gran Canaria,
> Islas Canarias - Espa?a.
> Tfn: +34 828 111 000 Ext: 340
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Hi All.
Thanks???, Now i can blind transfer calls with "#" key, but in
featuresmap said
>>blindxfer => #1 ; Blind transfer
>>disconnect => *0 ; Disconnect
>>automon => *1 ; One Touch Record
>>atxfer => *2 ; Attended transfer
but any key with * during call, (budgetone or x-ten)
produce this in console
Attempting native bridge of SIP/u0002-5fdd and SIP/u0001-eb16
And only "#" key its ok to transfer (not #1 how is stablished in
featuresmap).
any more configuration in asterisk? i dont know what more to do.
C?sar Garc?a.
Director de Sistemas, IdecNet S.A.
Centro de Gesti?n de Red.
Edificio IdecNet. C/Juan XXIII 44.
E-35004, Las Palmas de Gran Canaria,
Islas Canarias - Espa?a.
Tfn: +34 828 111 000 Ext: 340
Arunachala escribi?:> Hi,
>
> Please include "tT" options in your Dial statements in
extensions.conf.
>
> Example:
>
> extensions.conf
>
> [default]
>
> exten => 0000,1,Dial(SIP/u0001&SIP/u0004,20,tT)
> exten => _0XXX,1, Dial(SIP/u${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@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)
>>
>>--
>>
>>C?sar Garc?a.
>> Director de Sistemas, IdecNet S.A.
>> Centro de Gesti?n de Red.
>> Edificio IdecNet. C/Juan XXIII 44.
>> E-35004, Las Palmas de Gran Canaria,
>> Islas Canarias - Espa?a.
>> Tfn: +34 828 111 000 Ext: 340
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users@lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users