ok, thanks. i was beginning to suspect as much but was hoping to
limit the number of components in our configuration.
thanks,
gene
On Mon, Sep 27, 2010 at 11:13 AM, Kevin P. Fleming wrote:
> On 09/27/2010 11:02 AM, Eugene Oden wrote:
>> is there a trick to get asterisk (1.6.2.13) to propagate
On 09/27/2010 11:02 AM, Eugene Oden wrote:
> is there a trick to get asterisk (1.6.2.13) to propagate
> codec-changing sip reinvites when directrtpsetup=yes?
>
> i'm trying to route calls to a gateway without keeping asterisk in the
> rtp stream.
You are looking for a SIP proxy; Asterisk is not a
is there a trick to get asterisk (1.6.2.13) to propagate
codec-changing sip reinvites when directrtpsetup=yes?
i'm trying to route calls to a gateway without keeping asterisk in the
rtp stream.
the gateway is first routing the call to a media server. when
connecting the call to the downstream ca
