Discussion:
[SR-Users] No BYE from Called party
Yuriy Gorlichenko
2014-09-05 06:37:10 UTC
Permalink
Hello All. I have kamailio with provider connection (trunk)
When I call to external number through my provider call extablished Ok. But
when i try hangup call from external number no BYE sended to me. When I
hangup call from my kamailio (internal num) I send by to exteral number and
it respond me Ok so session if fully complete. I guess that BYE from
external number not recieves to me because I have wrong routing header
fields at my INVITe or ACK messages, but can not find any information what
what header must recieve info to external number where send BYE at hangup
or thomething like this.

This is my little dump for situation wherer I hangup from internal number
and BYE finished successfully:



IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599
E....< ***@.'.
...6........G.RINVITE sip:12345678900-x+0HtyjCAUtz+5FpPkU+***@public.gmane.org:5060 SIP/2.0
Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
Via: SIP/2.0/UDP my.kamailio.com:5068
;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
Max-Forwards: 70
From: <sip:TrunkNum-x+0HtyjCAUtz+5FpPkU+***@public.gmane.org>;tag=as5872f19e
To: <sip:12345678900-x+0HtyjCAUtz+5FpPkU+***@public.gmane.org:5068>
Contact:<TrunkNum-neSRGDJur/***@public.gmane.org:5068>
Call-ID: 42b819d45c08c9d304343bf976c5b405-Giij33B2sxDH/***@public.gmane.org:50600
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Thu, 04 Sep 2014 21:53:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 544
Proxy-Authorization: Digest username="TrunkNum", realm="my.provider.com",
nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P", uri="
sip:12345678900-x+0HtyjCAUtz+5FpPkU+***@public.gmane.org:5060", qop=auth, nc=00000001,
cnonce="3619116795", response="f5bc1d8125dd9e448d2e73764823adee",
algorithm=MD5

v=0
o=root 1022912010 1022912010 IN IP4 my.kamailio.com
s=Asterisk PBX 12.5.0
c=IN IP4 my.kamailio.com
t=0 0
a=ice-lite
m=audio 30032 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30033
a=ice-ufrag:3o8JrqkF
a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2
a=candidate:TgT1dfTnI3kBgWQ

IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882
E..... ./...6...
........b..SIP/2.0 200 OK
Via: SIP/2.0/UDP my.kamailio.com:5068;rport=5068;received=my.kamailio.com
;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
Record-Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
From: <sip:TrunkNum-x+0HtyjCAUtz+5FpPkU+***@public.gmane.org>;tag=as5872f19e
To: <sip:12345678900-x+0HtyjCAUtz+5FpPkU+***@public.gmane.org:5068>;tag=5rF0FNamQ99gH
Call-ID: 42b819d45c08c9d304343bf976c5b405-Giij33B2sxDH/***@public.gmane.org:50600
CSeq: 102 INVITE
Contact: <sip:12345678900-mSJsHTdNzfxda3jwGnjewCjTT/***@public.gmane.org:5060;transport=udp>
User-Agent: provider agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 746
X-provider agentOutboundGateway: sip:5574012345678900-***@public.gmane.org
X-provider agentOutboundCarrierID: 23705946361020
X-provider agentCarrierRate: 0.20180
X-provider agentCloudRate: 0.00300
;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip
s=FreeSWITCH
c=IN IP4 externail.number.end.ip
t=0 0
a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP
m=audio 23216 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ssrc:2990874569 cname:pRs5xP





IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596
***@..@.K\
...6........\`.ACK sip:12345678900-mSJsHTdNzfxda3jwGnjewCjTT/***@public.gmane.org:5060;transport=udp
SIP/2.0
Via: SIP/2.0/UDP my.kamailio.com:5068
;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600
Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
Max-Forwards: 70
From: <sip:TrunkNum-***@public.gmane.org>;tag=as5872f19e
To: <sip:12345678900-x+0HtyjCAUtz+5FpPkU+***@public.gmane.org:5068>;tag=5rF0FNamQ99gH
Call-ID: 42b819d45c08c9d304343bf976c5b405-Giij33B2sxDH/***@public.gmane.org:50600
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0













IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617
***@.KC
...6........q.ZBYE sip:12345678900-mSJsHTdNzfxda3jwGnjewCjTT/***@public.gmane.org:5060;transport=udp
SIP/2.0
Via: SIP/2.0/UDP my.kamailio.com:5068
;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
Max-Forwards: 70
From: <sip:TrunkNum-***@public.gmane.org>;tag=as5872f19e
To: <sip:12345678900-x+0HtyjCAUtz+5FpPkU+***@public.gmane.org:5068>;tag=5rF0FNamQ99gH
Call-ID: 42b819d45c08c9d304343bf976c5b405-Giij33B2sxDH/***@public.gmane.org:50600
CSeq: 103 BYE
User-Agent: Asterisk PBX 12.5.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568
E..T....-.676...
***@..SIP/2.0 200 OK
Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com
;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
From: <sip:TrunkNum-***@public.gmane.org>;tag=as5872f19e
To: <sip:12345678900-x+0HtyjCAUtz+5FpPkU+***@public.gmane.org:5068>;tag=5rF0FNamQ99gH
Call-ID: 42b819d45c08c9d304343bf976c5b405-Giij33B2sxDH/***@public.gmane.org:50600
CSeq: 103 BYE
User-Agent: provider agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0




Thanks for help.
Daniel-Constantin Mierla
2014-09-05 08:54:56 UTC
Permalink
Hello,

I noticed that the ACK is missing the Contact header -- not sure if
specs mention anything about being mandatory or not, but you can try to
get the contact there.

Cheers,
Daniel
Post by Yuriy Gorlichenko
Hello All. I have kamailio with provider connection (trunk)
When I call to external number through my provider call extablished
Ok. But when i try hangup call from external number no BYE sended to
me. When I hangup call from my kamailio (internal num) I send by to
exteral number and it respond me Ok so session if fully complete. I
guess that BYE from external number not recieves to me because I have
wrong routing header fields at my INVITe or ACK messages, but can not
find any information what what header must recieve info to external
number where send BYE at hangup or thomething like this.
This is my little dump for situation wherer I hangup from internal
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599
Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
Via: SIP/2.0/UDP
my.kamailio.com:5068;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Thu, 04 Sep 2014 21:53:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 544
Proxy-Authorization: Digest username="TrunkNum",
realm="my.provider.com <http://my.provider.com>",
nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P",
cnonce="3619116795", response="f5bc1d8125dd9e448d2e73764823adee",
algorithm=MD5
v=0
o=root 1022912010 1022912010 IN IP4 my.kamailio.com
<http://my.kamailio.com>
s=Asterisk PBX 12.5.0
c=IN IP4 my.kamailio.com <http://my.kamailio.com>
t=0 0
a=ice-lite
m=audio 30032 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30033
a=ice-ufrag:3o8JrqkF
a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2
a=candidate:TgT1dfTnI3kBgWQ
IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882
E..... ./...6...
........b..SIP/2.0 200 OK
Via: SIP/2.0/UDP
my.kamailio.com:5068;rport=5068;received=my.kamailio.com
<http://my.kamailio.com>;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
Record-Route: <sip:my.proider.com
<http://my.proider.com>;lr=on;ftag=as5872f19e>
Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
CSeq: 102 INVITE
User-Agent: provider agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 746
X-provider agentOutboundCarrierID: 23705946361020
X-provider agentCarrierRate: 0.20180
X-provider agentCloudRate: 0.00300
v=0
o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip
s=FreeSWITCH
c=IN IP4 externail.number.end.ip
t=0 0
a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP
m=audio 23216 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ssrc:2990874569 cname:pRs5xP
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596
...6........\`.ACK
Via: SIP/2.0/UDP
my.kamailio.com:5068;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600
Route: <sip:my.proider.com <http://my.proider.com>;lr=on;ftag=as5872f19e>
Max-Forwards: 70
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617
...6........q.ZBYE
Via: SIP/2.0/UDP
my.kamailio.com:5068;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
Route: <sip:my.proider.com <http://my.proider.com>;lr=on;ftag=as5872f19e>
Max-Forwards: 70
CSeq: 103 BYE
User-Agent: Asterisk PBX 12.5.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568
E..T....-.676...
Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com
<http://my.kamailio.com>;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
CSeq: 103 BYE
User-Agent: provider agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0
Thanks for help.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Yuriy Gorlichenko
2014-09-05 10:55:17 UTC
Permalink
RFC not specified Contack header at ACK... So anyway I already tried it
yesterday)) Unsuccessfull...
Post by Daniel-Constantin Mierla
Hello,
I noticed that the ACK is missing the Contact header -- not sure if specs
mention anything about being mandatory or not, but you can try to get the
contact there.
Cheers,
Daniel
Hello All. I have kamailio with provider connection (trunk)
When I call to external number through my provider call extablished Ok.
But when i try hangup call from external number no BYE sended to me. When I
hangup call from my kamailio (internal num) I send by to exteral number and
it respond me Ok so session if fully complete. I guess that BYE from
external number not recieves to me because I have wrong routing header
fields at my INVITe or ACK messages, but can not find any information what
what header must recieve info to external number where send BYE at hangup
or thomething like this.
This is my little dump for situation wherer I hangup from internal number
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599
Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
Via: SIP/2.0/UDP my.kamailio.com:5068
;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Thu, 04 Sep 2014 21:53:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 544
Proxy-Authorization: Digest username="TrunkNum", realm="my.provider.com",
nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P", uri="
cnonce="3619116795", response="f5bc1d8125dd9e448d2e73764823adee",
algorithm=MD5
v=0
o=root 1022912010 1022912010 IN IP4 my.kamailio.com
s=Asterisk PBX 12.5.0
c=IN IP4 my.kamailio.com
t=0 0
a=ice-lite
m=audio 30032 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30033
a=ice-ufrag:3o8JrqkF
a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2
a=candidate:TgT1dfTnI3kBgWQ
IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882
E..... ./...6...
........b..SIP/2.0 200 OK
Via: SIP/2.0/UDP my.kamailio.com:5068;rport=5068;received=my.kamailio.com
;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
Record-Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
CSeq: 102 INVITE
User-Agent: provider agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 746
X-provider agentOutboundCarrierID: 23705946361020
X-provider agentCarrierRate: 0.20180
X-provider agentCloudRate: 0.00300
;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip
s=FreeSWITCH
c=IN IP4 externail.number.end.ip
t=0 0
a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP
m=audio 23216 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ssrc:2990874569 cname:pRs5xP
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596
...6........\`.ACK
Via: SIP/2.0/UDP my.kamailio.com:5068
;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600
Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
Max-Forwards: 70
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617
...6........q.ZBYE
Via: SIP/2.0/UDP my.kamailio.com:5068
;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
Max-Forwards: 70
CSeq: 103 BYE
User-Agent: Asterisk PBX 12.5.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568
E..T....-.676...
Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com
;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
CSeq: 103 BYE
User-Agent: provider agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0
Thanks for help.
_______________________________________________
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Daniel-Constantin Mierla
2014-09-05 12:45:20 UTC
Permalink
Be sure you checked the two types of ack requests: hop-by-hop (for
negative replies, where the contact is not important at all) and
end-to-end (which is for a 200ok).

Also, even not required by rfc, some UA implementations can be broken.

Anyhow, if you tested and doesn't help, I would try to use
record_route() for ACK. If that doesn't help, you will need the help of
the provider to tell you why it doesn't send the BYE.

Cheers,
Daniel
Post by Yuriy Gorlichenko
RFC not specified Contack header at ACK... So anyway I already tried
it yesterday)) Unsuccessfull...
Hello,
I noticed that the ACK is missing the Contact header -- not sure
if specs mention anything about being mandatory or not, but you
can try to get the contact there.
Cheers,
Daniel
Post by Yuriy Gorlichenko
Hello All. I have kamailio with provider connection (trunk)
When I call to external number through my provider call
extablished Ok. But when i try hangup call from external number
no BYE sended to me. When I hangup call from my kamailio
(internal num) I send by to exteral number and it respond me Ok
so session if fully complete. I guess that BYE from external
number not recieves to me because I have wrong routing header
fields at my INVITe or ACK messages, but can not find any
information what what header must recieve info to external number
where send BYE at hangup or thomething like this.
This is my little dump for situation wherer I hangup from
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599
<sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
Via: SIP/2.0/UDP
my.kamailio.com:5068;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Thu, 04 Sep 2014 21:53:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 544
Proxy-Authorization: Digest username="TrunkNum",
realm="my.provider.com <http://my.provider.com>",
nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P",
nc=00000001, cnonce="3619116795",
response="f5bc1d8125dd9e448d2e73764823adee", algorithm=MD5
v=0
o=root 1022912010 1022912010 IN IP4 my.kamailio.com
<http://my.kamailio.com>
s=Asterisk PBX 12.5.0
c=IN IP4 my.kamailio.com <http://my.kamailio.com>
t=0 0
a=ice-lite
m=audio 30032 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30033
a=ice-ufrag:3o8JrqkF
a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2
a=candidate:TgT1dfTnI3kBgWQ
IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882
E..... ./...6...
........b..SIP/2.0 200 OK
Via: SIP/2.0/UDP
my.kamailio.com:5068;rport=5068;received=my.kamailio.com
<http://my.kamailio.com>;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
Record-Route: <sip:my.proider.com
<http://my.proider.com>;lr=on;ftag=as5872f19e>
<sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
CSeq: 102 INVITE
User-Agent: provider agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 746
X-provider agentOutboundCarrierID: 23705946361020
X-provider agentCarrierRate: 0.20180
X-provider agentCloudRate: 0.00300
v=0
o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip
s=FreeSWITCH
c=IN IP4 externail.number.end.ip
t=0 0
a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP
m=audio 23216 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ssrc:2990874569 cname:pRs5xP
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596
...6........\`.ACK
Via: SIP/2.0/UDP
my.kamailio.com:5068;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600
Route: <sip:my.proider.com
<http://my.proider.com>;lr=on;ftag=as5872f19e>
Max-Forwards: 70
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617
...6........q.ZBYE
Via: SIP/2.0/UDP
my.kamailio.com:5068;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
Route: <sip:my.proider.com
<http://my.proider.com>;lr=on;ftag=as5872f19e>
Max-Forwards: 70
CSeq: 103 BYE
User-Agent: Asterisk PBX 12.5.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568
E..T....-.676...
Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com
<http://my.kamailio.com>;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
CSeq: 103 BYE
User-Agent: provider agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0
Thanks for help.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 -http://www.asipto.com
Sep 22-25, Berlin, Germany
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Yuriy Gorlichenko
2014-09-09 12:25:38 UTC
Permalink
Hello, this is me again)

I added record_route header, that was no result....

I did some tests with my problem and have some results than confused me
very hard...

I registed my trunk from asterisk to provider directly. Do some calls. No
errors- allpackets sends and recieved perfectly. Rgen I catch logs off
calls from kamailio ans asterisk to same trunk on same porviser. I eq
results and was surprised - packets are the same (without sdp off course
and little things such as uac-agent and other)

Maby I missed something but now I cannot find any reason why call to trunk
not catches BYE from called party

I added my traces at attachement...

thanks for help
Post by Daniel-Constantin Mierla
Be sure you checked the two types of ack requests: hop-by-hop (for
negative replies, where the contact is not important at all) and end-to-end
(which is for a 200ok).
Also, even not required by rfc, some UA implementations can be broken.
Anyhow, if you tested and doesn't help, I would try to use record_route()
for ACK. If that doesn't help, you will need the help of the provider to
tell you why it doesn't send the BYE.
Cheers,
Daniel
RFC not specified Contack header at ACK... So anyway I already tried it
yesterday)) Unsuccessfull...
Post by Daniel-Constantin Mierla
Hello,
I noticed that the ACK is missing the Contact header -- not sure if specs
mention anything about being mandatory or not, but you can try to get the
contact there.
Cheers,
Daniel
Hello All. I have kamailio with provider connection (trunk)
When I call to external number through my provider call extablished Ok.
But when i try hangup call from external number no BYE sended to me. When I
hangup call from my kamailio (internal num) I send by to exteral number and
it respond me Ok so session if fully complete. I guess that BYE from
external number not recieves to me because I have wrong routing header
fields at my INVITe or ACK messages, but can not find any information what
what header must recieve info to external number where send BYE at hangup
or thomething like this.
This is my little dump for situation wherer I hangup from internal number
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599
Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
Via: SIP/2.0/UDP my.kamailio.com:5068
;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Thu, 04 Sep 2014 21:53:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 544
Proxy-Authorization: Digest username="TrunkNum", realm="my.provider.com",
nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P", uri="
cnonce="3619116795", response="f5bc1d8125dd9e448d2e73764823adee",
algorithm=MD5
v=0
o=root 1022912010 1022912010 IN IP4 my.kamailio.com
s=Asterisk PBX 12.5.0
c=IN IP4 my.kamailio.com
t=0 0
a=ice-lite
m=audio 30032 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30033
a=ice-ufrag:3o8JrqkF
a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2
a=candidate:TgT1dfTnI3kBgWQ
IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882
E..... ./...6...
........b..SIP/2.0 200 OK
Via: SIP/2.0/UDP my.kamailio.com:5068;rport=5068;received=my.kamailio.com
;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600
Record-Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
CSeq: 102 INVITE
User-Agent: provider agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 746
X-provider agentOutboundCarrierID: 23705946361020
X-provider agentCarrierRate: 0.20180
X-provider agentCloudRate: 0.00300
;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip
s=FreeSWITCH
c=IN IP4 externail.number.end.ip
t=0 0
a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP
m=audio 23216 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ssrc:2990874569 cname:pRs5xP
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596
...6........\`.ACK
Via: SIP/2.0/UDP my.kamailio.com:5068
;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600
Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
Max-Forwards: 70
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617
...6........q.ZBYE
Via: SIP/2.0/UDP my.kamailio.com:5068
;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
Route: <sip:my.proider.com;lr=on;ftag=as5872f19e>
Max-Forwards: 70
CSeq: 103 BYE
User-Agent: Asterisk PBX 12.5.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568
E..T....-.676...
Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com
;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
CSeq: 103 BYE
User-Agent: provider agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0
Thanks for help.
_______________________________________________
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
Loading...