asterisk users - Feb 2004 - INFO/DTMF retransmissions in * not absorbed?

I'm wondering whether people know if there could be a problem
with * receiving retransmissions of INFO/DTMF requests.

I'm trying to play DTMF via INFO to *. If it takes a 200 reply too 
long to come back, the request is retransmitted. Whenever this
happens, the IVR down in PSTN reports that the number sequence
is incorrect.

This makes me guess that * turns INFO retransmissions into new
DTMF digits on the PSTN part.

Does anybody have the same experience? Is it a known problem? 
Are there any patches?

Thanks,

-jiri

--
Jiri Kuthan            http://iptel.org/~jiri/

Hi Jiri,

Been there.  We switched from INFO to RFC2833 for this same reason.  
Take a look at:
http://bugs.digium.com/bug_view_page.php?bug_id=0001033

Not only retransmissions are affected but out of order packets too.  
This behaviour can be partly blamed on the RFC:

"In addition, the INFO method does not define additional mechanisms
for ensuring in-order delivery. While the CSeq header will be
incremented upon the transmission of new INFO messages, this should
not be used to determine the sequence of INFO information. This is
due to the fact that there could be gaps in the INFO message CSeq
count caused by a user agent sending re-INVITES or other SIP
messages. "

Regards,
Andres



Jiri Kuthan wrote:
>I'm wondering whether people know if there could be a problem
>with * receiving retransmissions of INFO/DTMF requests.
>
>I'm trying to play DTMF via INFO to *. If it takes a 200 reply too 
>long to come back, the request is retransmitted. Whenever this
>happens, the IVR down in PSTN reports that the number sequence
>is incorrect.
>
>This makes me guess that * turns INFO retransmissions into new
>DTMF digits on the PSTN part.
>
>Does anybody have the same experience? Is it a known problem? 
>Are there any patches?
>
>Thanks,
>
>-jiri
>
>--
>Jiri Kuthan            http://iptel.org/~jiri/ 
>
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>  
>

-- 
Andres
Network Admin
http://www.telesip.net

Put them in different [context] and use include => context in 
your dial plan.

-Heison

On Wed, Feb 25, 2004 at 01:43:37AM +0900, Isamar Maia
wrote:
> 
> I am trying to make some call routing...
> 
> I have the following rules in the same context:
> 
> ; 1st rule
> exten => _1800.,1,Dial(SIP/......)
> exten => _1800.,2,Congestion
> 
> ; 2nd rule
> exten => _1.,1,Dial(SIP/......)
> exten => _1.,2,Congestion
> 
> The problem is that some 1800 calls are still going to the second rule.
> What is the best way to accomplish that?
> 
> Thanks,
> 
> Isamar Maia
> 
> 
> 
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