[asterisk-users] DTMF between Asterisk servers.

[Al Baker]

Just a thought. A while back there was discussion about the merits of
having a product (in that case an O/S) with contracted vendor support
or relying solely on "list" support.
I note in the post below where one responder states
" It may also have been because less than 23 hours had elapsed...".

Different strokes for different folks.... But  "23 Hours" is loooong 
time in
the production world with no help on a TELCO problem. Just an observation on
how differently folks see things and what folk need to recognize before
they dump their NORTEL etc and jump into "open-source".

Mark Hamilton wrote:
> No, I tried calling the inbound DID to see if DTMF passes through. And most
> times it does, however, it's not being relayed to the Asterisk server 2, and
> then to the direct external phoneline.
>
> I tried changing all dtmfmodes for the sip peer, for the inbound DID
> provider, and it didn't work, even tried playing with canreinvite, etc.
>
> Hence why my desperate plea for help.
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Davies
> Sent: April 8, 2008 11:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DTMF between Asterisk servers.
>
> I believe that what you described should "just work" with the caveat
> that "dtmf=inband" is rarely the right thing to do over SIP, and is
> prone to all sorts of DTMF detection and debounce issues.
>
> I assume you've tried calling a POTS endpoint and listening to see if
> you get DTMF passed through?
>
> 1) You did not give a great deal of information about what the current
> situation was, or what investigations you've already tried, which is
> probably why no-one felt they could reply.
> 2) It may also have been because less than 23 hours had elapsed...
>
> Regards,
> Steve
>
> On 08/04/2008, Mark Hamilton <mark.h at cage151.com> wrote:
>   
>> I find it  hard to believe no one knows, so is it just plain no helping? J
>>
>> If someone would like to atleast point me in the right direction that will
>> deal specifically with what I'm asking, that would be appreciated too.
>>
>> Much thanks.
>>
>> From: Mark Hamilton [mailto:mark.h at cage151.com]
>>  Sent: April 7, 2008 11:48 AM
>>  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>>  Subject: DTMF between Asterisk servers.
>>
>> Hello,
>>
>> I'm a little confused on DTMF.
>>
>> A sip peer is registered on two Asterisk servers. No dtmfmode is set for
>> them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both
>> register on each other.
>>
>>
>>
>> A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the
>>     
> call
>   
>> is transferred to Asterisk 2:
>>
>>
>>     
> RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at 65.xx.xx.10,,t
> T,)
>   
>> Where 12351 accepts the call on Asterisk 2, and in some cases, that call
>>     
> is
>   
>> transferred out to a PSTN number, or wherever, but not within Asterisk
>> anymore via provider2, dtmf=rfc2833.
>>
>> When the call comes in, I'd like it to relay DTMF just dandy. How can I do
>> so?
>>
>> There is no NAT between the Asterisk servers or in front of them. However,
>> Asterisk2 has iptables which allows all UDP traffic  to/fro Asterisk1.
>>     
> When
>   
>> Asterisk2 transfers the call to external endpoints, there might be a LAN,
>> but relative ports are open on those LANs.
>>
>> Please help.
>>
>> Thanks in advance,
>>
>> Mark.
>>     
>
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