asterisk users - Jan 2006 - canreinvite always =no * no matter what we try :-(

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

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@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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--
"Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org"

Steve,
> The mission is to actually get a reinvite to work on the lan.
There isn't anything special to get this working... normally. I trust
you verified the traffic flow with a network monitor tool (tcpdump?),
correct? Does SIP debug give you any info (i.e., does it match the
right peer) -- you don't show if you allow reinvites globally? What
about the nat= setting?

Couple pointers I can give you to get you excited:
1) Reinvites work quite reliably, I use them between the PTSN gateway
and the end user's ATA, all the way across the Internet -- nicely
reduces latency.

2) If you use RFC2833 for DTMF you can issue an reinvite and still use
t/T for transfer. NOTE that you have to modify the source to make
asterisk reinvite even when it needs to listen to DTMFs. I give no
guarantees how well it will work for you but it does work.

See "AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1" in rtp.c.

3) Reinvites *can* work even if both ends are behind NAT. It really
depends on the NATing router and the ATA. Sipura's and good NAT
routers work, but I would not call it "reliable" -- it's really
pushing it a bit...

So if you really want to see why your Reinvites do not work, then you
probably will have to make your hands dirty and analyze where
ast_rtp_bridge() in rtp.c bails out. But since you are on a LAN it
makes the situation a lot easier.

--Luki

On Mon, 23 Jan 2006, Steve Gladden 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!

Remove from your Dial command all options that require Asterisk to hear 
the media stream.  (T, t etc)

Steve