reason shouldn't make any difference.

CSec should 'increase' or just be a new one from the invite.

I see that VIA branch have changed from invite and further messages and the bye one. I think that branch should stay the same as from the Invite.

mirko

On Wed, Nov 23, 2016 at 9:20 AM, devang nathwani <devang.nathwani31589@gmail.com> wrote:
Yes this is helping
but for the 'Contact' part in BYE header

below is the 'BYE' sip block from the working call, and it also not containing 'Contact:'
=========
2016/11/23 08:30:05.595033 UAC_IP:5060 -> PROXY_IP:5060
BYE sip:2648784@41.218.72.6:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP UAC_IP:5060;branch=z9hG4bK76fdsg00e0oh925na7l1cd0000g10.1
Call-ID: tlgfv877nn82btngv2vlbtagn9a799cn@SoftX3000
From: <sip:2787750@UAC_IP;user=phone>;tag=vf77f89n-CC-23
To: <sip:2648784@PROXY_IP;user=phone>;tag=yvXByKme39FKF
CSeq: 3 BYE
Reason: Q.850;cause=16;text="normal call clearing"
Max-Forwards: 69
Content-Length: 0
Route: <sip:PROXY_IP:5060;lr;ftag=vf77f89n-CC-23;did=b39.83362695>
=========

To compare above block with not working one, below is the block which is getting '481 Call Does Not Exist' from media
=========
2016/11/21 14:21:36.931816 UAC_IP:5060 -> PROXY_IP:5060
BYE sip:2648362@MEDIA_IP:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP UAC_IP:5060;branch=z9hG4bKnabcr510eo49dgoabt50.1
From: <sip:+4475642@UAC_IP:5060;user=phone>;tag=07001094154893
To: <sip:2648362@PROXY_IP:5060;user=phone>;tag=aDa8SH7vpgg5F
Max-Forwards: 69
CSeq: 448960898 BYE
Reason: Q.850;cause=16
Content-Length: 0
Route: <sip:PROXY_IP:5060;lr;ftag=07001094154893;did=9f6.51b0761>
=========

However, i find values of 'CSeq' and 'Reason' fields bit suspicious.

On Tue, Nov 22, 2016 at 9:09 PM, Mirko Brankovic <mirkobrankovic@gmail.com> wrote:
I think that your UAC is not adding Contact header to BYE message.
So try Adding :
Contact: <sip:UAC_IP:5060;transport=udp>
from the initial invite.

Also make sure that all VIA headers are there and unchanged

Hope it helps now :)

Mirko

On Tue, Nov 22, 2016 at 2:09 PM, devang nathwani <devang.nathwani31589@gmail.com> wrote:
Hello,

Now the tags are not changing any where in the call, here is the sip trace for the call
Yet freeswitch is sending '481 Call Does Not Exist'
Where could be issue?

On Fri, Nov 18, 2016 at 6:17 PM, Mirko Brankovic <mirkobrankovic@gmail.com> wrote:
HI,
I'm not sure if .17 is Freeswitch and who is 12 and wh 13 but I see that on first 480 no answer To:.... tag is changed:
  1. 2016/11/18 11:30:42.698223 11.23.16.13:5060 -> 11.23.16.12:5060
  2. SIP/2.0 480 NO_ANSWER
  3. Via: SIP/2.0/UDP 11.23.16.12:5060;branch=z9hG4bKlj0b84100gnjom3o1320.1
  4. From: <sip:+263733209456@11.23.16.12:5060;user=phone>;tag=09002861124617
  5. To: <sip:264812778555@11.23.16.13:5060;user=phone>;tag=f949601897706458a5166612fe67c373-1439
  6. CSeq: 986549121 INVITE
  7. Content-Length: 0
WHich I think you should avoid, changing tags. I think this might be the problem.

On Fri, Nov 18, 2016 at 1:07 PM, devang nathwani <devang.nathwani31589@gmail.com> wrote:
Hello,

Here are the sip traces of two legs
leg1 UAC -> proxy -> media
leg2 media -> proxy -> provider
from leg1,
when freeswitch(media) is sending back 480 temporarily unavailable after 183 session progress, UAC is sending ack and bye, now in return freeswitch is sending '481 Call Does Not Exist'

my question is why? why freeswitch is sending 481 in return of ack and bye?

here,
11.23.16.12 is UAC
11.23.16.13 is Opensips(Proxy)
11.23.16.16 is Freeswitch(Media)

Thanks,
Devang Nathwani

_________________________________________________________________________
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Regards,
Mirko

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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Official FreeSWITCH Sites
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Regards,
Mirko

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
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FreeSWITCH-users mailing list
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_________________________________________________________________________
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consulting@freeswitch.org
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Regards,
Mirko