[asterisk-users] Asterisk call transfers not working

[Carlos Rojas]

Hello,

That sound a tones problem, what do you seting, dtmf in your sip.conf?

Regards

On Mon, Oct 24, 2011 at 2:15 PM, Ramiro Paz <ramiro at masterline-logistics.com
> wrote:

> Hi everibody:
>
> Sorry, I want to relive this issue. I still have the problem, if somebody
> could help me will be appreciated. Tks.
>
> *Ramiro PAZ
> MASTERLINE LOGISTICS
> *
> **
> On Wed, Oct 19, 2011 at 3:25 PM, Ramiro Paz <
> ramiro at masterline-logistics.com> wrote:
>
>> Hi Danny, Warren:
>>
>> This is what I found in extensions_additional.conf:
>>
>> [from-internal-additional]
>> include => from-internal-additional-custom
>> include => app-dialvm
>> include => app-vmmain
>> include => app-recordings
>> include => app-callwaiting-cwoff
>> include => app-callwaiting-cwon
>> include => ext-group
>> include => grps
>> include => ext-queues
>> include => app-queue-toggle
>> include => app-calltrace
>> include => app-directory
>> include => app-echo-test
>> include => app-speakextennum
>> include => app-speakingclock
>> include => app-cf-busy-off
>> include => app-cf-busy-off-any
>> include => app-cf-busy-on
>> include => app-cf-off
>> include => app-cf-off-any
>> include => app-cf-on
>> include => app-cf-unavailable-off
>> include => app-cf-unavailable-on
>> include => app-cf-toggle
>> include => app-fmf-toggle
>> include => ext-findmefollow
>> include => fmgrps
>> include => app-userlogonoff
>> include => ext-local-confirm
>> include => findmefollow-ringallv2
>> include => app-pickup
>> include => app-zapbarge
>> include => app-chanspy
>> include => ext-test
>> include => ext-local
>> include => outbound-allroutes
>> exten => h,1,Hangup
>>
>> ; end of [from-internal-additional]
>>
>> There is nothing for [from-internal-custom]. I mean
>> extensions_custom.conf is empty.
>>
>> Just in case, Warren is right, we're using FXS/DAHDI endpoints. Thanks for
>> your time.
>>  *
>> Ramiro PAZ
>> **MASTERLINE LOGISTICS*
>>
>>
>> On Wed, Oct 19, 2011 at 12:59 PM, Warren Selby <wcselby at selbytech.com>wrote:
>>
>>> On Wed, Oct 19, 2011 at 11:28 AM, Danny Nicholas <danny at debsinc.com>wrote:
>>>
>>>> Or you could just add these lines to [from-internal-xfer]
>>>>
>>>> Exten => _X,1,Dial(SIP/${EXTEN},30,iKkTtt)****
>>>>
>>>> Exten => _XX,1,Dial(SIP/${EXTEN},30,iKkTt)****
>>>>
>>>> ** **
>>>>
>>>> If you have 3 or 4 digit extensions you would need these lines****
>>>>
>>>> Exten => _XXX,1,Dial(SIP/${EXTEN},30,iKkTtt)****
>>>>
>>>> Exten => _XXXX,1,Dial(SIP/${EXTEN},30,iKkTt)****
>>>>
>>>>
>>>>
>>> Except he's not sending to SIP endpoints, he's sending to FXS / DAHDI
>>> endpoints.  So the syntax would be a bit more specific based on which
>>> extension was being dialed and which port it was hooked up to on the card.
>>>
>>> --
>>> Thanks,
>>> --Warren Selby, dCAP
>>> http://www.SelbyTech.com <http://www.selbytech.com>
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>               http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111024/f7ad4085/attachment.htm>