Asterisk PBX Source Patch Needed

I need someone to patch asterisk source code to provide dial tone on E&M type signal methods - Currently Asterisk does not allow for a Dial Tone to be sent on E&M trunks - I prefer either a entry in [url removed, login to view] like dialtone=yes to indicate dial tone should OR should NOT be sent to device or (perfered) create a new type E&M or E&M Wink methods that does allow for dial tone such as E&M_DT and E&M_W_DT.

Whatever changes you make, I would like you to present the changes to the Asterisk Development Group so that it will appear in future releases of Asterisk. Please see instructions at [url removed, login to view] on how to do this.

A quick scan of the source reveals the effected file is chan_zap.c which can be downloaded from [url removed, login to view] - As far as I can tell, you might be able to simply copy sections in chan_zap.c that pertains to PRI and into the new E&M types, but I'm sure that not the best way nor the only changes that might be required. Please see source documentation at [url removed, login to view] .

I assume if you bid on this project that you do have knowledge of asterisk source and have the ability to complete this project as described above in a timely fashion. Please don't send "can" replies, they will be automatically deleted - place some words in you bid that show me you have read the above please.

Habilidades: Serviços de audio, Programação C, Linux

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Acerca do Empregador:
( 6 comentários ) Buena Park, United States

ID do Projeto: #78138

1 freelancer está oferecendo em média $100 para esse trabalho


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$100 USD in 5 dias
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