[Asterisk-Dev] The Almighty X-Lite DTMF Problem

[James Sizemore]

Another thing odd is I don't get phone to phone dtmf  using 
dtmfmode=rfc2833 on each sip extension.

/var/log/asterisk/messages:
Aug 15 18:30:01 WARNING[41998]: File rtp.c, Line 852 
(ast_rtp_senddigit): Don't know how to represent ''
Aug 15 18:30:01 WARNING[41998]: File rtp.c, Line 852 
(ast_rtp_senddigit): Don't know how to represent ''
Aug 15 18:30:01 WARNING[41998]: File rtp.c, Line 852 
(ast_rtp_senddigit): Don't know how to represent ''
Aug 15 18:30:01 WARNING[41998]: File rtp.c, Line 852 
(ast_rtp_senddigit): Don't know how to represent ''
Aug 15 18:30:04 WARNING[41998]: File rtp.c, Line 852 
(ast_rtp_senddigit): Don't know how to represent ''
Aug 15 18:30:04 WARNING[41998]: File rtp.c, Line 852 
(ast_rtp_senddigit): Don't know how to represent ''

So it looks like rtp is never getting any digits when two sip channels 
attempting native bridge. Look like dev/null
get all my key tones <smile>.  



James Sizemore wrote:

> When I try and pickup a SIP call from a pickup group  it now
> locks up Asterisk.  Can anyone else verify this new bug?
>
>
> Mark Spencer wrote:
>
>> Yes, it was merged yesterday.
>>
>> Mark
>>
>> On Fri, 15 Aug 2003, James Sizemore wrote:
>>
>>  
>>
>>> Do you know if this is in cvs yet?
>>>
>>> Chris Heiser wrote:
>>>
>>>   
>>>
>>>> Pete,
>>>>
>>>> Try this patch below...  I noticed that eStara's softphone has the 
>>>> same
>>>> problem as xten's softphone when it comes to DTMF.  Seems as though 
>>>> Asterisk
>>>> is not looking for the "end" bit per RFC2833.  So try this fix.  It 
>>>> should
>>>> do the trick (at least... it fixed mine).
>>>>
>>>> --Chris
>>>>
>>>> Index: rtp.c
>>>> ===================================================================
>>>> RCS file: /usr/cvsroot/asterisk/rtp.c,v
>>>> retrieving revision 1.22
>>>> diff -r1.22 rtp.c
>>>> 205a206
>>>>
>>>>
>>>>     
>>>>
>>>>>     unsigned int event_end;
>>>>>
>>>>>
>>>>>       
>>>>
>>>> 209a211,213
>>>>
>>>>
>>>>     
>>>>
>>>>>     event_end = ntohl(*((unsigned int *)(data)));
>>>>>     event_end <<= 8;
>>>>>     event_end >>= 24;
>>>>>
>>>>>
>>>>>       
>>>>
>>>> 224a229,234
>>>>
>>>>
>>>>     
>>>>
>>>>>     else if(event_end & 0x80)
>>>>>     {
>>>>>             f = send_dtmf(rtp);
>>>>>             resp = 0;
>>>>>     }
>>>>>
>>>>>
>>>>>
>>>>>       
>>>>
>>>>
>>>>     
>>>>
>>>>> -----Original Message-----
>>>>> From: asterisk-dev-admin at lists.digium.com [mailto:asterisk-dev-
>>>>> admin at lists.digium.com] On Behalf Of pgrace at fierymoon.com
>>>>> Sent: Tuesday, August 12, 2003 2:35 PM
>>>>> To: asterisk-dev at lists.digium.com
>>>>> Subject: [Asterisk-Dev] The Almighty X-Lite DTMF Problem
>>>>>
>>>>> Hey guys,
>>>>>
>>>>> I just was told by Rob at xten that the timestamp problem is fixed 
>>>>> in the
>>>>> rfc2833 implementation.  I'm still having the exact same problems 
>>>>> with
>>>>> voicemail(2) that I was before.  Can someone please un-resolve bug 
>>>>> 14 and
>>>>> maybe I can work with someone to help debug what's happening?
>>>>>
>>>>> Chris H, if you're still following this topic, fire me off an 
>>>>> e-mail if
>>>>> you want to see new debugs..
>>>>>
>>>>> Thanks,
>>>>> Pete (km-)
>>>>> _______________________________________________
>>>>> Asterisk-Dev mailing list
>>>>> Asterisk-Dev at lists.digium.com
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-dev
>>>>>
>>>>>
>>>>>       
>>>>
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>>>>
>>>>     
>>>
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>>>   
>>
>>
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>>  
>>
>
>
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