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

Sat Aug 11 05:18:27 MSD 2012

Its been said a million times on this ML for bugs to be posted to
jira,  further read the wiki on using git bisect to track down the
exact revision it broke in, that will greatly increase the chance of
the bug being taken care of asap.

~mitch

On Fri, Aug 10, 2012 at 1:43 AM, Johannes Jakob <lists.jj at googlemail.com> wrote:
> 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
>>
>>
>> ======================================================================================
>
>
>
> _________________________________________________________________________
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>
> 
> 
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