Michael Shuler wrote:
> Thanks, I know though, I paid for it to be done ;) And the whole
> community gets to benefit from it, guess that's my contribution back
> to the open source world that I use all the time to make money with
> :)
>
For whatever is worth.. THANKS :)
_
Olle E. Johansson wrote:
> Check out the latest CVS, Mark applied changes to the code in this
> area tonight. The rtp.c is changed, so the old patch in
> bugs.digium.com may not be necessary any more.
>
Yes, it is done..
BUT
Now I get MUCH higher values is the debug messages and can not
understa
on
> Sent: Sunday, March 14, 2004 2:12 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] ast_rtp_raw_write errors
> distorting sound on G729 passthrough
>
>
> Check out the latest CVS, Mark applied changes to the code in
> this area tonight.
> The rtp.c is chan
Check out the latest CVS, Mark applied changes to the code in this area tonight.
The rtp.c is changed, so the old patch in bugs.digium.com may not be necessary any
more.
/Olle
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Senad Jordanovic wrote:
I just tried that. Did you have to recompile * in order for it to
work?
Yes, you have to recompile and install again. You are changing the
source code.
Did you come up with error listed below while compiling??
rtp.c: In function `ast_r
>>
>> I just tried that. Did you have to recompile * in order for it to
>> work?
>>
>>
>>
> Yes, you have to recompile and install again. You are changing the
> source code.
Did you come up with error listed below while compiling??
rtp.c: In function `ast_rtp_write':
r
Senad Jordanovic wrote:
Andres wrote:
Michael Shuler wrote:
When I use reinvites everything works perfectly (so phoneA<-->phoneB
directly works fine). When I shut off reinvites
(phoneA<-->asterisk<-->phoneB) I get the following with PhoneA
initiating the call:
Mar 12 14:43:23 DEBUG[12092
Andres wrote:
> Michael Shuler wrote:
>
>> When I use reinvites everything works perfectly (so phoneA<-->phoneB
>> directly works fine). When I shut off reinvites
>> (phoneA<-->asterisk<-->phoneB) I get the following with PhoneA
>> initiating the call:
>>
>> Mar 12 14:43:23 DEBUG[1209277232]: rt
Michael Shuler wrote:
When I use reinvites everything works perfectly (so phoneA<-->phoneB
directly works fine). When I shut off reinvites
(phoneA<-->asterisk<-->phoneB) I get the following with PhoneA initiating
the call:
Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference
When I use reinvites everything works perfectly (so phoneA<-->phoneB
directly works fine). When I shut off reinvites
(phoneA<-->asterisk<-->phoneB) I get the following with PhoneA initiating
the call:
Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference
is 3360, ms is 440
Ma
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