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[Prometheus001 at gmx.net]

I have set up a FS behind a NAT.
Calling internal numbers, VM etc. works from a different network
(Caller's phone is also natted).
However if I call an external Gateway the call is terminated as soon as
the remote party lifts the handset. And Freeswitch is delivering
"SIP/2.0 500 Cannot Get IP Address for Media" to the caller and then
sends a BYE.
I can see that RTP packets are coming in from the external gateway from
port 26726 and they are sent to the callers phone before hangup, but no
RTP packets are coming from the callers phone and are send to the
external gateway.
But as I mentionned, calling internal number from this phone works. Also
the gateway works from another FS installation.

But what does this message mean in detail? Does he complain about the IP
adress of the caller or about the IP of the getaway, or does he complain
that he may not reach a certain port for the media?

Best regards
Peter


Here's the part of the FS log
====================
2008-11-21 20:49:27 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state()
Channel sofia/external/06XX16XXXX3@sip.qsc.de entering state [proceeding]
2008-11-21 20:49:27 [DEBUG] switch_ivr_bridge.c:292
audio_bridge_thread() sofia/internal/1000@my.domain.com receive message
[SWITCH_MESSAGE_INDICATE_RINGING]
2008-11-21 20:49:27 [DEBUG] switch_core_session.c:489
switch_core_session_perform_receive_message() Send signal
sofia/internal/1000@my.domain.com [BREAK]
2008-11-21 20:49:30 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state()
Channel sofia/external/06XX16XXXX3@sip.qsc.de entering state [ready]
2008-11-21 20:49:30 [DEBUG] switch_channel.c:1657
switch_channel_perform_mark_answered() Send signal
sofia/internal/1000@my.domain.com [BREAK]
2008-11-21 20:49:30 [NOTICE] sofia.c:2741 sofia_handle_sip_i_state()
Channel [sofia/external/06XX16XXXX3@sip.qsc.de] has been answered
2008-11-21 20:49:30 [DEBUG] switch_channel.c:1712
switch_channel_perform_answer() sofia/internal/1000@my.domain.com
receive message [SWITCH_MESSAGE_INDICATE_ANSWER]
2008-11-21 20:49:30 [DEBUG] mod_sofia.c:502 sofia_answer_channel() Local
SDP sofia/internal/1000@my.domain.com:
v=0
o=FreeSWITCH 1227284490 1227284492 IN IP4 my.domain.com
s=FreeSWITCH
c=IN IP4 my.domain.com
t=0 0
m=audio 12476 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
2008-11-21 20:49:30 [DEBUG] switch_core_session.c:489
switch_core_session_perform_receive_message() Send signal
sofia/internal/1000@my.domain.com [BREAK]
2008-11-21 20:49:30 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state()
Channel sofia/internal/1000@my.domain.com entering state [terminated]
2008-11-21 20:49:30 [NOTICE] sofia.c:2857 sofia_handle_sip_i_state()
Hangup sofia/internal/1000@my.domain.com [CS_EXECUTE]
[NORMAL_TEMPORARY_FAILURE]
2008-11-21 20:49:30 [NOTICE] switch_ivr_bridge.c:303
audio_bridge_thread() Channel [sofia/internal/1000@my.domain.com] has
been answered
2008-11-21 20:49:30 [DEBUG] switch_channel.c:1441
switch_channel_perform_hangup() Send signal
sofia/internal/1000@my.domain.com [KILL]
2008-11-21 20:49:30 [DEBUG] switch_core_session.c:779
switch_core_session_signal_state_change() Send signal
sofia/internal/1000@my.domain.com [BREAK]
2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:361
audio_bridge_thread() sofia/internal/1000@my.domain.com ending bridge by
request from read function
2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:436
audio_bridge_thread() BRIDGE THREAD DONE [sofia/internal/1000@my.domain.com]
2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:440
audio_bridge_thread() Send signal sofia/external/06XX16XXXX3@sip.qsc.de
[BREAK]
2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:913
switch_ivr_multi_threaded_bridge()
(sofia/external/06XX16XXXX3@sip.qsc.de) State Change CS_EXCHANGE_MEDIA
-> CS_RESET
2008-11-21 20:49:30 [DEBUG] switch_core_session.c:779
switch_core_session_signal_state_change() Send signal
sofia/external/06XX16XXXX3@sip.qsc.de [BREAK]
2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:433
switch_core_session_run() (sofia/internal/1000@my.domain.com) State
EXECUTE going to sleep
2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:375
switch_core_session_run() (sofia/internal/1000@my.domain.com) Running
State Change CS_HANGUP
2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:403
switch_core_session_run() (sofia/internal/1000@my.domain.com) State HANGUP
2008-11-21 20:49:30 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel
sofia/internal/1000@my.domain.com hanging up, cause:
NORMAL_TEMPORARY_FAILURE
2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup() sofia/internal/1000@my.domain.com
Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:403
switch_core_session_run() (sofia/internal/1000@my.domain.com) State
HANGUP going to sleep
2008-11-21 20:49:30 [DEBUG] switch_core_session.c:911
switch_core_session_thread() Session 15
(sofia/internal/1000@my.domain.com) Locked, Waiting on external entities
2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:410
audio_bridge_thread() sofia/external/06XX16XXXX3@sip.qsc.de receive
message [SWITCH_MESSAGE_INDICATE_UNBRIDGE]
2008-11-21 20:49:30 [DEBUG] switch_core_session.c:489
switch_core_session_perform_receive_message() Send signal
sofia/external/06XX16XXXX3@sip.qsc.de [BREAK]
2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:436
audio_bridge_thread() BRIDGE THREAD DONE
[sofia/external/06XX16XXXX3@sip.qsc.de]
2008-11-21 20:49:30 [DEBUG] switch_ivr_bridge.c:440
audio_bridge_thread() Send signal sofia/internal/1000@my.domain.com [BREAK]
2008-11-21 20:49:30 [NOTICE] switch_ivr_bridge.c:472
audio_bridge_on_exchange_media() Hangup
sofia/external/06XX16XXXX3@sip.qsc.de [CS_RESET] [NORMAL_CLEARING]
2008-11-21 20:49:30 [DEBUG] switch_channel.c:1441
switch_channel_perform_hangup() Send signal
sofia/external/06XX16XXXX3@sip.qsc.de [KILL]
2008-11-21 20:49:30 [DEBUG] switch_core_session.c:779
switch_core_session_signal_state_change() Send signal
sofia/external/06XX16XXXX3@sip.qsc.de [BREAK]
2008-11-21 20:49:30 [NOTICE] switch_core_session.c:929
switch_core_session_thread() Session 15
(sofia/internal/1000@my.domain.com) Ended
2008-11-21 20:49:30 [NOTICE] switch_core_session.c:931
switch_core_session_thread() Close Channel
sofia/internal/1000@my.domain.com [CS_HANGUP]
2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:436
switch_core_session_run() (sofia/external/06XX16XXXX3@sip.qsc.de) State
EXCHANGE_MEDIA going to sleep
2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:375
switch_core_session_run() (sofia/external/06XX16XXXX3@sip.qsc.de)
Running State Change CS_HANGUP
2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:403
switch_core_session_run() (sofia/external/06XX16XXXX3@sip.qsc.de) State
HANGUP
2008-11-21 20:49:30 [DEBUG] mod_sofia.c:253 sofia_on_hangup()
sofia/external/06XX16XXXX3@sip.qsc.de Overriding SIP cause 480 with 500
from the other leg
2008-11-21 20:49:30 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel
sofia/external/06XX16XXXX3@sip.qsc.de hanging up, cause: NORMAL_CLEARING
2008-11-21 20:49:30 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending
BYE to sofia/external/06XX16XXXX3@sip.qsc.de
2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup() sofia/external/06XX16XXXX3@sip.qsc.de
Standard HANGUP, cause: NORMAL_CLEARING
2008-11-21 20:49:30 [DEBUG] switch_core_state_machine.c:403
switch_core_session_run() (sofia/external/06XX16XXXX3@sip.qsc.de) State
HANGUP going to sleep
2008-11-21 20:49:30 [DEBUG] switch_core_session.c:911
switch_core_session_thread() Session 16
(sofia/external/06XX16XXXX3@sip.qsc.de) Locked, Waiting on external entities
2008-11-21 20:49:30 [NOTICE] switch_core_session.c:929
switch_core_session_thread() Session 16
(sofia/external/06XX16XXXX3@sip.qsc.de) Ended
2008-11-21 20:49:30 [NOTICE] switch_core_session.c:931
switch_core_session_thread() Close Channel
sofia/external/06XX16XXXX3@sip.qsc.de [CS_HANGUP]



Network-Scan on the FS machine:
===========================================
U 213.148.136.2:5060 -> 10.101.0.201:5080
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP
195.xxx.xxx.2xx:5080;branch=z9hG4bKZ0pFrtcctSS4m;rport=5080.
Call-ID: 1820daf3-325d-122c-e4a8-001d096ac1c9.
From: "Extension
1000"<sip:07141xxxxxx@sip.qsc.de;transport=udp>;tag=pevpy0DUNSQpS.
To: <sip:06XX16XXXX3@sip.qsc.de>;tag=207dc7d8.
CSeq: 107513988 INVITE.
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER.
Contact: <sip:213.148.136.2:5060;user=phone>.
Content-Length: 0.
.

#
U 213.148.136.2:5060 -> 10.101.0.201:5080
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
195.xxx.xxx.2xx:5080;branch=z9hG4bKZ0pFrtcctSS4m;rport=5080.
Call-ID: 1820daf3-325d-122c-e4a8-001d096ac1c9.
From: "Extension
1000"<sip:07141xxxxxx@sip.qsc.de;transport=udp>;tag=pevpy0DUNSQpS.
To: <sip:06XX16XXXX3@sip.qsc.de>;tag=207dc7d8.
CSeq: 107513988 INVITE.
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER.
Supported: 100rel,replaces,timer,precondition,histinfo.
Contact: <sip:213.148.136.2:5060;user=phone>.
Content-Length: 221.
Content-Type: application/sdp.
.
v=0.
o=HuaweiSoftX3000 10118984 10118985 IN IP4 213.148.136.2.
s=Sip Call.
c=IN IP4 213.148.136.2.
t=0 0.
m=audio 26726 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.
a=fmtp:101 0-15.

#
U 10.101.0.201:5080 -> 213.148.136.2:5060
ACK sip:213.148.136.2:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 195.xxx.xxx.2xx:5080;rport;branch=z9hG4bK09F8SNXFQ2FQg.
Max-Forwards: 70.
From: "Extension 1000"
<sip:07141xxxxxx@sip.qsc.de;transport=udp>;tag=pevpy0DUNSQpS.
To: <sip:06XX16XXXX3@sip.qsc.de>;tag=207dc7d8.
Call-ID: 1820daf3-325d-122c-e4a8-001d096ac1c9.
CSeq: 107513988 ACK.
Contact: <sip:07141xxxxxx@195.xxx.xxx.2xx:5080;transport=udp>.
Proxy-Authorization: Digest username="07141xxxxxx", realm="qsc.de",
nonce="492693bf3670f4927f1ad7f6310fb22b65e10c17",
cnonce="GCLZVTJdEiyo5AAdCWrByQ", algorithm=MD5,
uri="sip:06XX16XXXX3@sip.qsc.de",
response="79c35e38c30eb7c25a92f9b90526abd5", qop=auth, nc=00000001.
Content-Length: 0.
.

#
U 10.101.0.201:5060 -> 217.XX.XX.189:2048
*SIP/2.0 500 Cannot Get IP Address for Media.*
Via: SIP/2.0/UDP 217.XX.XX.189:2048;branch=z9hG4bK-kgkuy3yyoq8a;rport=2048.
From: "1000" <sip:1000@my.domain.com>;tag=44eejmbuny.
To: <sip:06XX16XXXX3@my.domain.com;user=phone>;tag=56v0XXNSBSHNc.
Call-ID: 3c2718517d92-pifrb7rpiz4f.
CSeq: 2 INVITE.
Contact: <sip:mod_sofia@my.domain.com:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10438.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk.
Content-Length: 0.


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[Prometheus001 at gmx.net]

Nobody has an idea?


Best regards
Peter

Peter P GMX schrieb:

[torstein.knutsen at gm...]

I don't have a clue, but I've seen the same as you.

I did setup an fresh install of freeswitch.trunk (10473) on official IP, and one user (also on official IP). And got the same error 500 Cannot Get IP Addess for media .... vhen dialling trough external sip_profile to a gateway.
I don't have any solution, but I'm also interested in what excactly this error tells me ..

Best regards
Torstein



On Sun, Nov 23, 2008 at 3:24 PM, Peter P GMX <Prometheus001@gmx.net (Prometheus001@gmx.net)> wrote:

[brian at freeswitch.org]

Ok this is why.. DO NOT put hostnames in the external RTP ip or the
sip ip's those are ONLY IP addresses.

/b

On Nov 21, 2008, at 2:00 PM, Peter P GMX wrote:

[anthony.minessale at g...]

Do you always ask a question then send a follow up to your own email asking us to
hurry up and answer you? This is 2 weeks in a row iirc.
It's somewhat rude. This is the weekend after all.


On Sun, Nov 23, 2008 at 6:30 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[Prometheus001 at gmx.net]

Thanks for the hint.
I had indeed set the rtp-ip to the fqdn of the server in internal.xml,
as this way I got rid of one way audio for internal calls.
Calls through external gateways didn't work though due to the error we
are talking about here.

Now I added a
<param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
entry to internal.xml and external.xml in the SIP profiles and now the
RTP stream via internal and external profile is o.k.

I added a new chapter "Freeswitch behind NAT" to the wiki, where I
describe my settings:
http://wiki.freeswitch.org/wiki/NAT_Traversal

Best regards
Peter


Brian West schrieb: