[Freeswitch-users] Call Diversion/CF (302 Moved Temporarily) not working

Tue Aug 7 20:03:54 MSD 2012

Hi,

My setup looks like this:

User A (1001), User B (1002) and User C (1003) are registered to
13.13.13.66 via OpenSIPs Proxy 222.222.222.222.
User B activated unconditional call forwarding on his phone to User C.
Now User A (or any external caller) calls User B.

Until last week User A would then be able to talk to User C - the
expected behaviour.
On Saturday, I updated the freeswitch from version 9fe08675a1 to
d1c3f910a6 to get and test some of Steve's fax changes.
Since this upgrade, Call-Forwarding / Call Diversion is broken on a
level, that I don't understand.

User A: INVITE 1002 to B's phone
=> User B: 302 Moved Temporarily (Contact: <sip:1003 at sip.isp.net;user=phone>)
=> FS: ACK
=> FS: INVITE 1003 to *B*'s phone!
=> User B: 404 Not Found
=> FS: ACK
=> User A gets signalled the 404 Not Found.

packet trace below.

I'm aware of
http://jira.freeswitch.org/browse/FS-724
http://jira.freeswitch.org/browse/FS-821

but those didn't help me solve my issue:

internal.xml:		<param name="apply-nat-acl" value="nat.auto"/>
internal.xml:		<param name="aggressive-nat-detection" value="false"/>

I added aggressive-nat-detection=false yesterday, before it wasn't in
the config at all.
I also tried removing the nat.auto acl temporarily, but that didn't help either.

Yesterday afternoon I updated to git c3de9637af, did several
bootstrappings, cleans and makes... no change.

So today I tried to downgrade to 9fe08675a1, the version I had running
until Friday and that was working fine... well, it doesn't now. So I
can't even _prove_ it was working before :(

Can somebody please point me in the right direction?

Any hint in the right direction is much appreciated!

Thanks!

======================================================================================

SIP Trace (tcpdump):

16:38:58.950321 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472
E....[ .>..;^...
	.......D.fINVITE sip:1002 at 137.137.137.245:7390;line=ydwzxzpg SIP/2.0
Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285>
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
Via: SIP/2.0/UDP
13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB
Max-Forwards: 28
From: "User A" <sip:1001 at sip.isp.net>;tag=3r8K7r5Hppe6N
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>
Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
CSeq: 31839105 INVITE
Contact: <sip:mod_sofia at 13.13.13.66:5060>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 442
P-Key-Flags: keys="3"
X-AUTH-IP: 137.137.137.245
X-FS-Support: update_display,send_info
Remote-Party-ID: "User A"
<sip:1001 at sip.isp.net>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1344322582 1344322583 IN IP4 13.13.13.66
s=FreeSWITCH
c=IN IP4 13.13.13.66
t=0 0
m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 13
a=rtpmap:98 G7221/32000
a=fmtp:98 bitrate=48000
a=rtpmap:99 G7221/16000
a=fmtp:99 bitrate=32000
a=rtpmap:100 iLBC/8000
a=fmtp:100 mode=30
a=rtpmap:101 teleph
16:38:58.950328 IP 222.222.222.222 > 172.16.12.210: udp
E....[..>.4.^...
	..one-event/8000
a=fmtp:101 0-16
a=ptime:20
m=video 27936 RTP/AVP 34 102
a=rtpmap:34 H263/90000
a=rtpmap:102 H264/90000

16:38:58.979148 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 623
E..... at .@.
.
	..^........w8$SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
Via: SIP/2.0/UDP
13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB
Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285>
From: "User A" <sip:1001 at sip.isp.net>;tag=3r8K7r5Hppe6N
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>;tag=mzey6xfxhz
Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
CSeq: 31839105 INVITE
Contact: <sip:1003 at sip.isp.net;user=phone>
Diversion: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>;reason="unconditional"
Content-Length: 0

16:38:58.979954 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 358
.^.... at .>.
	.......n.fACK sip:1002 at 137.137.137.245:7390;line=ydwzxzpg SIP/2.0
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
From: "User A" <sip:1001 at sip.isp.net>;tag=3r8K7r5Hppe6N
Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>;tag=mzey6xfxhz
CSeq: 31839105 ACK
Max-Forwards: 70
Content-Length: 0

16:38:58.985405 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472
E....\ .>..:^...
	.......ZgIINVITE sip:1003 at 137.137.137.245:7390;user=phone SIP/2.0
Record-Route: <sip:222.222.222.222;lr;ftag=411c9KpNKZ4rH;did=e81.20343ac4>
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
Via: SIP/2.0/UDP
13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ
Max-Forwards: 28
From: "User A" <sip:1001 at sip.isp.net>;tag=411c9KpNKZ4rH
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>
Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
CSeq: 31839105 INVITE
Contact: <sip:mod_sofia at 13.13.13.66:5060>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 466
P-Key-Flags: keys="3"
X-AUTH-IP: 137.137.137.245
X-FS-Support: update_display,send_info
Remote-Party-ID: "User A"
<sip:1001 at sip.isp.net>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1344322582 1344322584 IN IP4 13.13.13.66
s=FreeSWITCH
c=IN IP4 13.13.13.66
t=0 0
m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101
a=rtpmap:98 G7221/32000
a=fmtp:98 bitrate=48000
a=rtpmap:99 G7221/16000
a=fmtp:99 bitrate=32000
a=rtpmap:100 iLBC/8000
a=fmtp:100 mode=30
a=rtpmap:101 telephone-e
16:38:58.985409 IP 222.222.222.222 > 172.16.12.210: udp
E....\..>.4.^...
	..vent/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
m=video 27936 RTP/AVP 34 102
a=rtpmap:34 H263/90000
a=rtpmap:102 H264/90000

16:38:59.031218 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 551
E..C.. at .@..7
	..^......../..SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
Via: SIP/2.0/UDP
13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ
From: "User A" <sip:1001 at sip.isp.net>;tag=411c9KpNKZ4rH
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>
Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
CSeq: 31839105 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Length: 0

16:38:59.031922 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 341
E..q.. at .>...^...
	.......].mACK sip:1003 at 137.137.137.245:7390;user=phone SIP/2.0
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
From: "User A" <sip:1001 at sip.isp.net>;tag=411c9KpNKZ4rH
Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>
CSeq: 31839105 ACK
Max-Forwards: 70
Content-Length: 0

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