[asterisk-users] Problem with dtmf and voice mail

[Dovid Bender]

Never really messed with a at h. Sorry :(
----- Original Message ----- 
From: "Paul A Brown" <paul at fowlmere.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Monday, August 14, 2006 5:33 PM
Subject: Re: [asterisk-users] Problem with dtmf and voice mail


> Hi Dovid,
>
> I can't see how to easily do that in *@home :-(
>
> Any ideas?
>
> Paul
> ----- Original Message ----- 
> From: "Dovid Bender" <asteriskusers at dovid.net>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users at lists.digium.com>
> Sent: Sunday, August 13, 2006 3:44 PM
> Subject: Re: [asterisk-users] Problem with dtmf and voice mail
>
>
>>I had a problem with asterisk real time that if in the general section of 
>>sip.conf i was using one form of dtmf and in the real time i set another 
>>the dtmf would not work for the first while (dont remember exactly how 
>>long). It could be a bug in asterisk. Try making the dtmf in the general 
>>section and under that phones setting in sip.conf (or real time) the same 
>>and see what happens.
>>
>> Dovid
>>
>> ----- Original Message ----- 
>> From: "Paul A Brown" <paul at fowlmere.com>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> <asterisk-users at lists.digium.com>
>> Sent: Friday, August 11, 2006 6:25 PM
>> Subject: Re: [asterisk-users] Problem with dtmf and voice mail
>>
>>
>>> Cheers Dean
>>>
>>> In extensions config I tried
>>>
>>> inbound
>>> rfc2833
>>> auto
>>> info
>>>
>>> I saved and rebooted phone after each but the problem seemed to stay. 
>>> Could it be a phone issue?
>>>
>>> Thanks
>>>
>>> Paul
>>>
>>> ----- Original Message ----- 
>>> From: "Dean Collins" <Dean at collins.net.pr>
>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>>> <asterisk-users at lists.digium.com>
>>> Sent: Friday, August 11, 2006 5:45 PM
>>> Subject: RE: [asterisk-users] Problem with dtmf and voice mail
>>>
>>>
>>> Hi Paul, Happy Friday back.
>>>
>>>
>>> In the config of the extension change the dtmf=XXX
>>>
>>> Basically there are three ways dtmf can be transmitted by a sip handset,
>>> choose another or search the voip-info for the options and you'll solve
>>> your problem pretty quickly.
>>>
>>> Re: sipgate....sorry cant help, you'll need to provide more info.
>>>
>>>
>>>
>>> Cheers,
>>>
>>> Dean
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
>>>> bounces at lists.digium.com] On Behalf Of Paul A Brown
>>>> Sent: Friday, 11 August 2006 9:30 AM
>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>> Subject: [asterisk-users] Problem with dtmf and voice mail
>>>>
>>>> Hi Guys,
>>>>
>>>> Happy Friday
>>>>
>>>> I have 2 problems....
>>>>
>>>> I run Asterisk at home with some Cisco 7960's
>>>>
>>>> 1) DTMF - When I dial a number on the 7960 it works fine. However if I
>>> dial
>>>> a number that asks 'Dial 1 for this and 2 for that' and I hit 1 or 2
>>> (or
>>>> whatever0 the other end acts as though nothing is heard. Any ideas?
>>>>
>>>> 2) Voicemail - I use a company called sipgate for my internal route.
>>> When
>>>> someone calls from outsied the call never goes to vmail. However if I
>>> dial
>>>> from ext to ext it does...
>>>>
>>>> Any ideas?
>>>>
>>>> Thanks
>>>>
>>>> Paul
>>>>
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>>
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