[asterisk-users] Problem with dtmf and voice mail

[Dovid Bender]

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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