[Asterisk-Users] sip native bridge vs. sip reinvite

[Jeremy Jones]

Hi,

I'm trying to get rtp media streams to run between endpoints rather than
through my * server, and I think I'm getting something wrong.  I have an
AS5300 speaking both h323 (for a different voip system I run) and sip
for *.  Dial-peers on the as5300 differentiate inbound from pstn to
different chunks of DID numbers between h323 and sip.  I'm testing with
xlite on a PC.

So here's what I have:

Outbound trunks are defined in my extensions.conf that send _9whatever
to SIP/pstn_gw/${EXTEN}.

In sip.conf I have two friends, one for my xlite softphone, one for
pstn_gw:

[2085551212]
type=friend
username=2085551212
secret=1234
host=dynamic
canreinvite=yes
disallow=all
allow=ulaw
context=testme
mailbox=5551212
callerid="Jeremy Jones" <2085551212>

[pstn_gw]
type=friend
username=pstn_gw
disallow=all
allow=ulaw
context=default
canreinvite=yes
host=10.0.0.201

I can place a call from the PSTN to 5551212 successfully, and I can
place calls from xlite to the PSTN successfully.  But in either case I
always see two sip channels active on *, and the endpoints (as5300 &
xlite) are sending their rtp via *.  Here's what I see when I place a
call from xlite to:


*CLI>     -- Executing Prefix("SIP/2085551212-f04d", "9") in new stack
    -- Prepended prefix, new extension is 93532533
    -- Executing Dial("SIP/20825551212-f04d", "SIP/pstn_gw/93532533") in
new stack
    -- Called pstn_gw/93532533
    -- SIP/pstn_gw-85a0 is making progress passing it to
SIP/2085551212-f04d
    -- SIP/pstn_gw-85a0 answered SIP/2085551212-f04d
    -- Attempting native bridge of SIP/2085551212-f04d and
SIP/pstn_gw-85a0

*CLI>
*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter
Format
10.0.0.201       93532533    02e2e09e167  00103/00651  00000ms  0000ms
ULAW
10.0.0.100     2082874602    E0541F6D-81  00102/03763  00000ms  0000ms
ULAW
2 active SIP channel(s)
*CLI>

(I have a Prefix rule for outbound 'cuz this is a system for residential
users, and the as5300 has dial-peers that need a 9 prefix...)

The output in * is similar for inbound from PSTN to xlite.

I can send output from sip debug if that'd help.

Thanks,
Jeremy Jones
Network Nerd
WestCom, LLC