[Freeswitch-users] SIP Overlap support?

[Brian West]

We have <param name="rfc2833-pt" value="101"/> on the profile and you  
to set this.

So you can set it to 96 if needed.  But you shouldn't have to do that  
if they say 96 and we say 101 they should be listening on 101 from us  
and we should be listening on 96 from them... thats why its called an  
RTP map.

/b

On Oct 31, 2009, at 10:48 AM, Patrick List wrote:

>> On Oct 24, 2009, at 8:13 AM, Dennis wrote:
>>
>>> ok, as written, i come back after some tests with fs and a thomson
>>> cirpack.
>
> No idea if this is useful as I'm a noob with fs. If not please excuse
> the noise. In the past Asterisk to work properly with Cirpack needed  
> the
> following patch:
>
> diff -uNr asterisk-1.4.19.org/main/rtp.c asterisk-1.4.19/main/rtp.c
> --- asterisk-1.4.19.org/main/rtp.c	2007-10-08 22:06:33.000000000 +0200
> +++ asterisk-1.4.19/main/rtp.c	2007-11-11 13:12:28.000000000 +0100
> @@ -1383,6 +1383,7 @@
>  	[34] = {1, AST_FORMAT_H263},
>  	[103] = {1, AST_FORMAT_H263_PLUS},
>  	[97] = {1, AST_FORMAT_ILBC},
> +	[96] = {0, AST_RTP_DTMF},
>  	[99] = {1, AST_FORMAT_H264},
>  	[101] = {0, AST_RTP_DTMF},
>  	[110] = {1, AST_FORMAT_SPEEX},
>
> Maybe this helps.
>
> Regards,
> Patrick
>
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