[Asterisk-Users] canreinvite always =no * no matter what we try :-(

[Steve Gladden]

> How are you testing if asterisk is in the media path?

Two ways:

One phone on a hub with ethereal on a laptop and watching the rtp
packets, pretty obvious that asterisk is staying in the media path.
and that the rtp i not coming from the other phone.

Way two, in the middle of an active/established call unplugging the
ethernet cable from the asterisk box
audio instantly dies on both phones when this occurs.
plug asterisk box back into it's ethernet termination....
audio comes right back.

Seems odd that these reinvites are supposed to magically occur
(from what I gather) and it only happens when the sun is shining
and everything is just right...

I'd like a way to force it or KNOW that it should be occuring
versus just expecting it to 'possibly' occur automatically if
all conditions are met and automatically detected.

Or maybe I have this all worng :-)

Thanks!
Steve









> please turn on all the debug, warning, error etc messages in the
> console, see logger.conf, then type sip peer <peer1> debug and sip
> peer <peer2> debug to see the SIP messages.
>
> How are you testing if asterisk is in the media path?
>
> Regards
>
> On 1/23/06, Steve Gladden <Asterisk at michiganbroadband.com> wrote:
>> been testing with a rather simple setup.
>>
>> The mission is to actually get a reinvite to work on the lan.
>>
>> I am trying with two sipura phones G.711 codec forced on both
>> both on the lan no nat no fancy options suchs as tT or H
>>
>> No matter what we do asterisk hangs on to the media path, how
>> in the world do I get a reinvite to work where the media path
>> is actually handled by the two phones on the lan?
>>
>> Any pointers greatly appreciated!
>>
>> Steve
>>
>>
>> Pretty simple extensions, on lan no nat
>>
>> <sip.conf>
>> [4785]
>>
>> type=friend
>> username=4785
>> secret=test
>> host=dynamic
>> canreinvite=yes
>>
>> [4786]
>>
>> type=friend
>> username=4786
>> secret=tesst
>> host=dynamic
>> canreinvite=yes
>>
>> <extensions.conf>
>> exten => 4785,1,Dial(SIP/4785,66)
>> exten => 4785,3,hangup
>>
>> exten => 4786,1,Dial(SIP/4786,66)
>> exten => 4786,3,hangup
>>
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>
>
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