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

Fri Aug 10 12:43:14 MSD 2012

Hi guys,

so really nobody got an idea about that one?
It's driving me nuts ;)

_Can_ this be some kind of dialplan fuckup?

Regards,

Johannes 

-- 
Johannes Jakob
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

On Tuesday, 7. August 2012 at 18:03, Johannes Jakob wrote:

> 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 (mailto: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 (mailto: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 (mailto:1001 at sip.isp.net)>;tag=3r8K7r5Hppe6N
> To: <sip:1002 at 137.137.137.245 (mailto: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 (mailto: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 (mailto: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 (mailto:1001 at sip.isp.net)>;tag=3r8K7r5Hppe6N
> To: <sip:1002 at 137.137.137.245 (mailto: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 (mailto:1003 at sip.isp.net);user=phone>
> Diversion: <sip:1002 at 137.137.137.245 (mailto: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 (mailto: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 (mailto:1001 at sip.isp.net)>;tag=3r8K7r5Hppe6N
> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
> To: <sip:1002 at 137.137.137.245 (mailto: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 (mailto: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 (mailto:1001 at sip.isp.net)>;tag=411c9KpNKZ4rH
> To: <sip:1002 at 137.137.137.245 (mailto: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 (mailto: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 (mailto: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 (mailto:1001 at sip.isp.net)>;tag=411c9KpNKZ4rH
> To: <sip:1002 at 137.137.137.245 (mailto: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 (mailto: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 (mailto:1001 at sip.isp.net)>;tag=411c9KpNKZ4rH
> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>
> CSeq: 31839105 ACK
> Max-Forwards: 70
> Content-Length: 0
> 
> 
> ====================================================================================== 