[asterisk-users] Asterisk call transfers not working

[Carlos Rojas]

Hello,

Did you to use, dtmfmode = info?

Regards

On Mon, Oct 24, 2011 at 3:13 PM, Ramiro Paz <ramiro at masterline-logistics.com
> wrote:

> 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>
>>>>>
>>>>>
>>>>> --
>>>>> _____________________________________________________________________
>>>>> -- 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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>>>>>
>>>>
>>>>
>>>
>>> --
>>> _____________________________________________________________________
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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
>>
>
>
> --
> _____________________________________________________________________
> -- 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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