1 Nov
2014
1 Nov
'14
9:39 p.m.
Hi, I have kamilio-4.2 and rtpengine running on the same machine.I have this scenario:
softphone --> Kamailio with Rtpengine --> AsteriskThe softphone initiates the call,
it is sent to the Asterisk. I can see the SDPs being re-written with the new IP/Ports
provided by rtpengine:
Invite from Kamailio to Asterisk200 Ok from Kamailio to Softphone
However, I take a signaling/media capture on the server where the kamailio/rtpengine are
running and see the RTP coming from both endpoints (softphone and asterisk) to the correct
ports but there is no packets coming out from the proxy to either direction.
I see these 2 lines on the rtpengine log and make me think that something prevents the
rtpengine to stream out to the 2 endpoints:
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] Write error on RTP socket
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] Final packet stats:
this is the log from the rtpengine:
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: Got valid command from 127.0.0.1:46686:
offer - { "sdp": "v=0#015#012o=- 314474204173996793 6663595733338147866 IN
IP4 99.153.161.148#015#012s=-#015#012c=IN IP4 99.153.161.148#015#012t=0 0#015#012m=audio
16386 RTP/AVP 114 0 8 101#015#012a=rtpmap:114 ilbc/8000#015#012a=fmtp:114
mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8 pcma/8000#015#012a=rtpmap:101
telephone-event/8000#015#012a=fmtp:101 0-15#015#012", "direction": [
"pub", "pub" ], "replace": [ "origin",
"session-connection" ], "call-id":
"0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [
"IP4", " ...
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: ... 99.153.161.148" ],
"from-tag": "73rH43p60NeKm", "command": "offer" }
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd]
Creating new call
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd]
Opened ports 35022..35023 for media relay
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd]
Opened ports 35038..35039 for media relay
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd]
Returning to SIP proxy: d3:sdp529:v=0#015#012o=- 314474204173996793 6663595733338147866 IN
IP4 54.173.60.13#015#012s=-#015#012c=IN IP4 54.173.60.13#015#012t=0
0#015#012a=ice-lite#015#012m=audio 35022 RTP/AVP 114 0 8 101#015#012a=rtpmap:114
ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8
pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=sendrecv#015#012a=rtcp:35023#015#012a=ice-ufrag:oPm3I7OE#015#012a=ice-pwd:WjZnh7dnhfdkfoXDoqDbnCLLr7N7#015#012a=candidate:NZ1IXXQHYxOora9j
1 UDP 2130706431 54.173.60.13 35022 typ host#015#012a=candidate: ...
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] ...
NZ1IXXQHYxOora9j 2 UDP 2130706430 54.173.60.13 35023 typ host#015#0126:result2:oke
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: Got valid command from 127.0.0.1:46686:
answer - { "sdp": "v=0#015#012o=root 439969655 439969655 IN IP4
205.10.248.62#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 205.10.248.62#015#012t=0
0#015#012m=audio 11142 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101
telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=silenceSupp:off - - -
-#015#012a=ptime:20#015#012a=sendrecv#015#012", "direction": [
"pub", "pub" ], "replace": [ "origin",
"session-connection" ], "call-id":
"0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [
"IP4", "205.10.248.62" ], "fr ...
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: ... om-tag":
"73rH43p60NeKm", "to-tag": "as57e7b8e3",
"command": "answer" }
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd]
Returning to SIP proxy: d3:sdp500:v=0#015#012o=root 439969655 439969655 IN IP4
54.173.60.13#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 54.173.60.13#015#012t=0
0#015#012a=ice-lite#015#012m=audio 35038 RTP/AVP 0 101#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-16#015#012a=silenceSupp:off - - -
-#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:35039#015#012a=ice-ufrag:cBL9SIw9#015#012a=ice-pwd:QHw7VHp83nMsLpD8ZoSqtDxGchN4#015#012a=candidate:NZ1IXXQHYxOora9j
1 UDP 2130706431 54.173.60.13 35038 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP
213070 ...
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] ...
6430 54.173.60.13 35039 typ host#015#0126:result2:oke
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] Write error on RTP socket
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] Final packet stats:
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] --- Tag 'as57e7b8e3', created 0:00 ago, in dialogue with
'73rH43p60NeKm'
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] ------ Media #1, port 35022 <> 205.10.248.62:11142, 0 p, 0 b, 0 e
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] ------ Media #1, port 35023 <> 205.10.248.62:11143 (RTCP), 0 p, 0 b, 0 e
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] --- Tag '73rH43p60NeKm', created 0:00 ago, in dialogue with
'as57e7b8e3'
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] ------ Media #1, port 35038 <> 99.153.161.148:16386, 0 p, 0 b, 1 e
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] ------ Media #1, port 35039 <> 99.153.161.148:16387 (RTCP), 0 p, 0 b, 0 e
Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] Got valid command from 127.0.0.1:46686: delete - { "call-id":
"0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [
"IP4", "205.10.248.62" ], "from-tag":
"as57e7b8e3", "command": "delete" }
Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] [0866b358-dc9c-1232-1399-3767db69b8dd] Call-ID to delete not found
Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port
35038] Returning to SIP proxy: d7:warning38:Call-ID not found or tags didn't
match6:result2:oke
this is the log from kamailio (showing only the lines that include the word rtpengine):
Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: INFO: <script>: mylog: TO_FS:
Starting rtpengine session. Invite
Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:request_route=[TO_FS]
c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=441 a=25
n=rtpengine_offer
Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine_funcs.c:148]:
check_content_type(): type <application/sdp> found valid
Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]:
rtpp_function_call(): proxy reply: d3:sdp529:v=0#015#012o=- 314474204173996793
6663595733338147866 IN IP4 54.173.60.13#015#012s=-#015#012c=IN IP4 54.173.60.13#015#012t=0
0#015#012a=ice-lite#015#012m=audio 35022 RTP/AVP 114 0 8 101#015#012a=rtpmap:114
ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8
pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=sendrecv#015#012a=rtcp:35023#015#012a=ice-ufrag:oPm3I7OE#015#012a=ice-pwd:WjZnh7dnhfdkfoXDoqDbnCLLr7N7#015#012a=candidate:NZ1IXXQHYxOora9j
1 UDP 2130706431 54.173.60.13 35022 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP
2130706430 54.173.60.13 35023 typ host#015#0126:result2:oke
Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: INFO: <script>: mylog: Starting
rtpengine session. Answer
Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:onreply_route=[TO_FS]
c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=476 a=25
n=rtpengine_answer
Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine_funcs.c:148]:
check_content_type(): type <application/sdp> found valid
Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]:
rtpp_function_call(): proxy reply: d3:sdp500:v=0#015#012o=root 439969655 439969655 IN IP4
54.173.60.13#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 54.173.60.13#015#012t=0
0#015#012a=ice-lite#015#012m=audio 35038 RTP/AVP 0 101#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-16#015#012a=silenceSupp:off - - -
-#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:35039#015#012a=ice-ufrag:cBL9SIw9#015#012a=ice-pwd:QHw7VHp83nMsLpD8ZoSqtDxGchN4#015#012a=candidate:NZ1IXXQHYxOora9j
1 UDP 2130706431 54.173.60.13 35038 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP
2130706430 54.173.60.13 35039 typ host#015#0126:result2:oke
Nov 1 18:59:36 ip-10-0-2-68 ./kamailio[28001]: exec: ***
cfgtrace:request_route=[LOOSE_ROUTE]
c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=488 a=24
n=rtpengine_delete
Nov 1 18:59:36 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]:
rtpp_function_call(): proxy reply: d7:warning38:Call-ID not found or tags didn't
match6:result2:oke
3 Nov
3 Nov
3:59 p.m.
On 11/01/14 15:39, Juan Perez wrote:
Hi, I have kamilio-4.2 and rtpengine running on the
same machine.
I have this scenario:
softphone --> Kamailio with Rtpengine --> Asterisk
The softphone initiates the call, it is sent to the Asterisk. I can see
the SDPs being re-written with the new IP/Ports provided by rtpengine:
Invite from Kamailio to Asterisk
200 Ok from Kamailio to Softphone
However, I take a signaling/media capture on the server where the
kamailio/rtpengine are running and see the RTP coming from both
endpoints (softphone and asterisk) to the correct ports but there is no
packets coming out from the proxy to either direction.
I see these 2 lines on the rtpengine log and make me think that
something prevents the rtpengine to stream out to the 2 endpoints:
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]:
[0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Write error on RTP socket
Seeing as you're using the "direction" options, can you double check
that the local IP addresses that you've configured at the command line
are actually addresses bound to local interfaces on the machine?
cheers
4:17 p.m.
thank you Richard, yes the IP is local to the machine:
./rtpengine --interface=pub/<PUBLIC_IP> --interface=priv/10.0.2.68
--listen-ng=127.0.0.1:7722--timeout=30 --port-min=35000 --port-max=65000 --log-level=7
--log-facility=daemon
The PUBLIC_IP is a NAT that the machine has, it is a virtual machine (Amazon), so it is
not configured on any interface in that machine.
But the Private one it is configured on the eth0:
[root@ip-10-0-2-68]# ifconfig
eth0 Link encap:Ethernet HWaddr 12:23:49:EF:3A:53
inet addr:10.0.2.68 Bcast:10.0.2.255 Mask:255.255.255.0
inet6 addr: fe80::1023:49ff:feef:3a53/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:9001 Metric:1
RX packets:161973 errors:0 dropped:0 overruns:0 frame:0
TX packets:102009 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:117022408 (111.6 MiB) TX bytes:20614133 (19.6 MiB)
Interrupt:247
This is how kamailio is setup to communicate with rtpengine and it is the only line I have
manually configured for that module in the kamailio config file, everything else is by
default
:
# ----- rtpengine params -----
modparam("rtpengine", "rtpengine_sock",
"udp:127.0.0.1:7722")
This is the line I have on the Route section to forward the INVITE to the Asterisk:
rtpengine_offer("direction=pub direction=pub replace-origin
replace-session-connection");
The whole Route section is this:
route[TO_FS] {
# here we load the Asterisk GWs that will be used to send the calls out.
t_on_reply("TO_FS");
t_on_failure("TO_FS");
$var(result) = load_gws(10);
rtpengine_offer("direction=pub direction=pub replace-origin
replace-session-connection");
xlog("L_INFO","mylog: TO_FS: Call received. Loading LCR_GRP
10\n");
if (!load_gws(10)) {
xlog("L_INFO","mylog: TO_FS: After, GW_URI_AVP:
$avp(i:709).\n");
sl_send_reply("503", "Unable to load destination
gateways");
xlog("L_INFO","mylog: TO_FS: Destination GWs load section
failed!. Load_GW function.\n");
exit;
}
if(!next_gw()){
xlog("L_INFO","mylog: TO_FS: After, GW_URI_AVP:
$avp(i:709).\n");
xlog("L_INFO","mylog: TO_FS: Destination GWs load section
failed!. Next_GW function.\n");
sl_send_reply("503", "Unable to find a gateway");
exit;
}
xlog("L_INFO","mylog: TO_FS: Destination GWs load section
OK.\n");
if (!t_relay()) {
xlog("L_INFO","mylog: TO_FS. T_Relay failed. Method
[$rm].\n");
sl_reply_error();
}
exit;
}
And this is the line I setup when I manage the Reply from the Asterisk:
onreply_route[TO_FS] {
xlog("L_INFO","mylog: OnReply Route TO_FS.\n");
if (has_body("application/sdp")) {
xlog("L_INFO","mylog: Starting rtpengine session.
Answer\n");
rtpengine_answer("direction=pub direction=pub replace-origin
replace-session-connection");
}
exit;
}
On Monday, November 3, 2014 8:59 AM, Richard Fuchs <rfuchs(a)sipwise.com> wrote:
On 11/01/14 15:39, Juan Perez wrote:
Hi, I have kamilio-4.2 and rtpengine running on the
same machine.
I have this scenario:
softphone --> Kamailio with Rtpengine --> Asterisk
The softphone initiates the call, it is sent to the Asterisk. I can see
the SDPs being re-written with the new IP/Ports provided by rtpengine:
Invite from Kamailio to Asterisk
200 Ok from Kamailio to Softphone
However, I take a signaling/media capture on the server where the
kamailio/rtpengine are running and see the RTP coming from both
endpoints (softphone and asterisk) to the correct ports but there is no
packets coming out from the proxy to either direction.
I see these 2 lines on the rtpengine log and make me think that
something prevents the rtpengine to stream out to the 2 endpoints:
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]:
[0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Write error on RTP socket
Seeing as you're using the "direction" options, can you double check
that the local IP addresses that you've configured at the command line
are actually addresses bound to local interfaces on the machine?
cheers
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
4:21 p.m.
On 11/03/14 09:17, Juan Perez wrote:
thank you Richard, yes the IP is local to the
machine:
./rtpengine --interface=pub/<PUBLIC_IP> --interface=priv/10.0.2.68
--listen-ng=127.0.0.1:7722--timeout=30 --port-min=35000 --port-max=65000
--log-level=7 --log-facility=daemon
The PUBLIC_IP is a NAT that the machine has, it is a virtual machine
(Amazon), so it is not configured on any interface in that machine.
There's your problem. If the address isn't configured on the machine,
rtpengine cannot use it as source address when sending packets out. If
RTP packets must be sent with one address as source, but another address
must appear in SDP bodies, use this format:
--interface=pub/<RTP SOURCE ADDRESS>!<SDP ADDRESS>
cheers
6:33 p.m.
thank you Richard,that was it, at least 1 side is working, the other side I think is
having issues with the Record-Route having the private IP in it.I know it is out of
context , but do you know how to make Kamailio to use the Public IP instead?txs a lot!jp
On Monday, November 3, 2014 9:17 AM, Juan Perez <juan_perez_2014(a)yahoo.com>
wrote:
thank you Richard, yes the IP is local to the machine:
./rtpengine --interface=pub/<PUBLIC_IP> --interface=priv/10.0.2.68
--listen-ng=127.0.0.1:7722--timeout=30 --port-min=35000 --port-max=65000 --log-level=7
--log-facility=daemon
The PUBLIC_IP is a NAT that the machine has, it is a virtual machine (Amazon), so it is
not configured on any interface in that machine.
But the Private one it is configured on the eth0:
[root@ip-10-0-2-68]# ifconfig
eth0 Link encap:Ethernet HWaddr 12:23:49:EF:3A:53
inet addr:10.0.2.68 Bcast:10.0.2.255 Mask:255.255.255.0
inet6 addr: fe80::1023:49ff:feef:3a53/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:9001 Metric:1
RX packets:161973 errors:0 dropped:0 overruns:0 frame:0
TX packets:102009 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:117022408 (111.6 MiB) TX bytes:20614133 (19.6 MiB)
Interrupt:247
This is how kamailio is setup to communicate with rtpengine and it is the only line I have
manually configured for that module in the kamailio config file, everything else is by
default
:
# ----- rtpengine params -----
modparam("rtpengine", "rtpengine_sock",
"udp:127.0.0.1:7722")
This is the line I have on the Route section to forward the INVITE to the Asterisk:
rtpengine_offer("direction=pub direction=pub replace-origin
replace-session-connection");
The whole Route section is this:
route[TO_FS] {
# here we load the Asterisk GWs that will be used to send the calls out.
t_on_reply("TO_FS");
t_on_failure("TO_FS");
$var(result) = load_gws(10);
rtpengine_offer("direction=pub direction=pub replace-origin
replace-session-connection");
xlog("L_INFO","mylog: TO_FS: Call received. Loading LCR_GRP
10\n");
if (!load_gws(10)) {
xlog("L_INFO","mylog: TO_FS: After, GW_URI_AVP:
$avp(i:709).\n");
sl_send_reply("503", "Unable to load destination
gateways");
xlog("L_INFO","mylog: TO_FS: Destination GWs load section
failed!. Load_GW function.\n");
exit;
}
if(!next_gw()){
xlog("L_INFO","mylog: TO_FS: After, GW_URI_AVP:
$avp(i:709).\n");
xlog("L_INFO","mylog: TO_FS: Destination GWs load section
failed!. Next_GW function.\n");
sl_send_reply("503", "Unable to find a gateway");
exit;
}
xlog("L_INFO","mylog: TO_FS: Destination GWs load section
OK.\n");
if (!t_relay()) {
xlog("L_INFO","mylog: TO_FS. T_Relay failed. Method
[$rm].\n");
sl_reply_error();
}
exit;
}
And this is the line I setup when I manage the Reply from the Asterisk:
onreply_route[TO_FS] {
xlog("L_INFO","mylog: OnReply Route TO_FS.\n");
if (has_body("application/sdp")) {
xlog("L_INFO","mylog: Starting rtpengine session.
Answer\n");
rtpengine_answer("direction=pub direction=pub replace-origin
replace-session-connection");
}
exit;
}
On Monday, November 3, 2014 8:59 AM, Richard Fuchs <rfuchs(a)sipwise.com> wrote:
On 11/01/14 15:39, Juan Perez wrote:
Hi, I have kamilio-4.2 and rtpengine running on the
same machine.
I have this scenario:
softphone --> Kamailio with Rtpengine --> Asterisk
The softphone initiates the call, it is sent to the Asterisk. I can see
the SDPs being re-written with the new IP/Ports provided by rtpengine:
Invite from Kamailio to Asterisk
200 Ok from Kamailio to Softphone
However, I take a signaling/media capture on the server where the
kamailio/rtpengine are running and see the RTP coming from both
endpoints (softphone and asterisk) to the correct ports but there is no
packets coming out from the proxy to either direction.
I see these 2 lines on the rtpengine log and make me think that
something prevents the rtpengine to stream out to the 2 endpoints:
Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]:
[0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Write error on RTP socket
Seeing as you're using the "direction" options, can you double check
that the local IP addresses that you've configured at the command line
are actually addresses bound to local interfaces on the machine?
cheers
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Juan Perez -
Richard Fuchs