[asterisk-users] Asterisk call transfers not working

[Ramiro Paz]

Hi Carlos:

I'm not using SIP endpoints, the problem is with FXS/DAHDI endpoints. Anyway
I could see I'm using RFC2833 for SIP extensions. Tks.

Regards,

*Ramiro PAZ
MASTERLINE LOGISTICS

***
On Mon, Oct 24, 2011 at 3:33 PM, Carlos Rojas <crt.rojas at gmail.com> wrote:

> 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>
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
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>>>
>>>
>>
>> --
>> _____________________________________________________________________
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>
>
> --
> _____________________________________________________________________
> -- 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
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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