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
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