Russell Bryant wrote:
> On Aug 11, 2008, at 12:04 PM, SIP wrote:
>
>
>> SIP wrote:
>>
>>> When calling from our SIP proxy through Asterisk to the PSTN
>>> provider,
>>> we support reINVITES which tend to work seamlessly.
>>>
>>> However, when creating a call file which essentially connect
On Aug 11, 2008, at 12:04 PM, SIP wrote:
> SIP wrote:
>> When calling from our SIP proxy through Asterisk to the PSTN
>> provider,
>> we support reINVITES which tend to work seamlessly.
>>
>> However, when creating a call file which essentially connects a call
>> from the SIP proxy to the SIP p
SIP wrote:
> When calling from our SIP proxy through Asterisk to the PSTN provider,
> we support reINVITES which tend to work seamlessly.
>
> However, when creating a call file which essentially connects a call
> from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP
> media path
When calling from our SIP proxy through Asterisk to the PSTN provider,
we support reINVITES which tend to work seamlessly.
However, when creating a call file which essentially connects a call
from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP
media path. I understand that th