Re: [SR-Users] No BYE from Called party

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

On 05/09/14 12:55, Yuriy Gorlichenko wrote:

RFC not specified Contack header at ACK... So anyway I already tried it yesterday)) Unsuccessfull...

2014-09-05 12:54 GMT+04:00 Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com>:

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


On 05/09/14 08:37, Yuriy Gorlichenko wrote:

...

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@my.provider.com:5060
<http://sip:12345678900@my.provider.com: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@my.provider.com
<mailto:sip%3ATrunkNum@my.provider.com>>;tag=as5872f19e
To: <sip:12345678900@my.provider.com:5068
<http://sip:12345678900@my.provider.com:5068>>
Contact:<TrunkNum@my.kamailio.com:5068
<http://TrunkNum@my.kamailio.com:5068>>
Call-ID:
42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600
<mailto:42b819d45c08c9d304343bf976c5b405@my.client.internal.ip: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 <http://my.provider.com>",
nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P",
uri="sip:12345678900@my.provider.com:5060
<http://sip:12345678900@my.provider.com:5060>", qop=auth,
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>
Record-Route:
<sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on>
From: <sip:TrunkNum@my.provider.com
<mailto:sip%3ATrunkNum@my.provider.com>>;tag=as5872f19e
To: <sip:12345678900@my.provider.com:5068
<http://sip:12345678900@my.provider.com:5068>>;tag=5rF0FNamQ99gH
Call-ID:
42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600
<mailto:42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600>
CSeq: 102 INVITE
Contact: <sip:12345678900@externail.number.end.ip: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@62.93.147.149
<mailto:sip%3A5574012345678900@62.93.147.149>
X-provider agentOutboundCarrierID: 23705946361020
X-provider agentCarrierRate: 0.20180
X-provider agentCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060@my.provider.com
<mailto:sip%3A5060@my.provider.com>>;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
E..p.@..@.K\ <mailto:E..p.@..@.K%5C>
...6........\`.ACK
sip:12345678900@externail.number.end.ip: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
<http://my.proider.com>;lr=on;ftag=as5872f19e>
Max-Forwards: 70
From: <sip:TrunkNum@sip.callsion.com
<mailto:sip%3ATrunkNum@sip.callsion.com>>;tag=as5872f19e
To: <sip:12345678900@my.provider.com:5068
<http://sip:12345678900@my.provider.com:5068>>;tag=5rF0FNamQ99gH
Call-ID:
42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600
<mailto:42b819d45c08c9d304343bf976c5b405@my.client.internal.ip: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
E....D..@.KC <mailto:E....D..@.KC>
...6........q.ZBYE
sip:12345678900@externail.number.end.ip: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
<http://my.proider.com>;lr=on;ftag=as5872f19e>
Max-Forwards: 70
From: <sip:TrunkNum@sip.callsion.com
<mailto:sip%3ATrunkNum@sip.callsion.com>>;tag=as5872f19e
To: <sip:12345678900@my.provider.com:5068
<http://sip:12345678900@my.provider.com:5068>>;tag=5rF0FNamQ99gH
Call-ID:
42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600
<mailto:42b819d45c08c9d304343bf976c5b405@my.client.internal.ip: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
<http://my.kamailio.com>;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0
Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600
From: <sip:TrunkNum@sip.callsion.com
<mailto:sip%3ATrunkNum@sip.callsion.com>>;tag=as5872f19e
To: <sip:12345678900@my.provider.com:5068
<http://sip:12345678900@my.provider.com:5068>>;tag=5rF0FNamQ99gH
Call-ID:
42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600
<mailto:42b819d45c08c9d304343bf976c5b405@my.client.internal.ip: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
sr-users@lists.sip-router.org  <mailto:sr-users@lists.sip-router.org>
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
sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users