<div dir="ltr"><span style="font-family:arial,sans-serif;font-size:13.3333339691162px">Hello, this is me again)</span><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><span style="font-family:arial,sans-serif;font-size:13.3333339691162px">I added record_route header, that was no result....</span><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><span style="font-family:arial,sans-serif;font-size:13.3333339691162px">I did some tests with my problem and have some results than confused me very hard...</span><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><span style="font-family:arial,sans-serif;font-size:13.3333339691162px">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)</span><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><span style="font-family:arial,sans-serif;font-size:13.3333339691162px">Maby I missed something but now I cannot find any reason why call to trunk not catches BYE from called party</span><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><span style="font-family:arial,sans-serif;font-size:13.3333339691162px">I added my traces at attachement...</span><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><br style="font-family:arial,sans-serif;font-size:13.3333339691162px"><span style="font-family:arial,sans-serif;font-size:13.3333339691162px">thanks for help</span><br><div class="gmail_extra"><br><div class="gmail_quote">2014-09-05 16:45 GMT+04:00 Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
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).<br>
<br>
Also, even not required by rfc, some UA implementations can be
broken.<br>
<br>
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.<br>
<br>
Cheers,<br>
Daniel<div><div><br>
<br>
<div>On 05/09/14 12:55, Yuriy Gorlichenko
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">RFC not specified Contack header at ACK... So
anyway I already tried it yesterday)) Unsuccessfull...</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">2014-09-05 12:54 GMT+04:00
Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> Hello,<br>
<br>
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.<br>
<br>
Cheers,<br>
Daniel
<div>
<div><br>
<br>
<div>On 05/09/14 08:37, Yuriy Gorlichenko wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
<div>
<div>
<div dir="ltr">Hello All. I have kamailio with
provider connection (trunk)<br>
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. <br>
<br>
This is my little dump for situation wherer I
hangup from internal number and BYE finished
successfully:<br>
<br>
<br>
<br>
<div>
<div>IP my.kamailio.com.5068 >
my.proider.com.5060: UDP, length 1599</div>
<div>E....< .@.'.</div>
<div>...6........G.RINVITE <a href="http://sip:12345678900@my.provider.com:5060" target="_blank">sip:12345678900@my.provider.com:5060</a>
SIP/2.0</div>
<div>Record-Route: <a><sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on></a></div>
<div>Via: SIP/2.0/UDP
my.kamailio.com:5068;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2</div>
<div>Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600</div>
<div>Max-Forwards: 70</div>
<div>From: <<a href="mailto:sip%3ATrunkNum@my.provider.com" target="_blank">sip:TrunkNum@my.provider.com</a>>;tag=as5872f19e</div>
<div>To: <<a href="http://sip:12345678900@my.provider.com:5068" target="_blank">sip:12345678900@my.provider.com:5068</a>></div>
<div>Contact:<<a href="http://TrunkNum@my.kamailio.com:5068" target="_blank">TrunkNum@my.kamailio.com:5068</a>></div>
<div>Call-ID: <a href="mailto:42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600" target="_blank">42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600</a></div>
<div>CSeq: 102 INVITE</div>
<div>User-Agent: Asterisk PBX 12.5.0</div>
<div>Date: Thu, 04 Sep 2014 21:53:13 GMT</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE</div>
<div>Supported: replaces, timer</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 544</div>
<div>Proxy-Authorization: Digest
username="TrunkNum", realm="<a href="http://my.provider.com" target="_blank">my.provider.com</a>",
nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P",
uri="<a href="http://sip:12345678900@my.provider.com:5060" target="_blank">sip:12345678900@my.provider.com:5060</a>",
qop=auth, nc=00000001, cnonce="3619116795",
response="f5bc1d8125dd9e448d2e73764823adee",
algorithm=MD5</div>
<div><br>
</div>
<div>v=0</div>
<div>o=root 1022912010 1022912010 IN IP4 <a href="http://my.kamailio.com" target="_blank">my.kamailio.com</a></div>
<div>s=Asterisk PBX 12.5.0</div>
<div>c=IN IP4 <a href="http://my.kamailio.com" target="_blank">my.kamailio.com</a></div>
<div>t=0 0</div>
<div>a=ice-lite</div>
<div>m=audio 30032 RTP/AVP 0 3 8 101</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:3 GSM/8000</div>
<div>a=rtpmap:8 PCMA/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div>
<div>a=ptime:20</div>
<div>a=maxptime:150</div>
<div>a=sendrecv</div>
<div>a=rtcp:30033</div>
<div>a=ice-ufrag:3o8JrqkF</div>
<div>a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2</div>
<div>a=candidate:TgT1dfTnI3kBgWQ</div>
<div><br>
</div>
<div>IP my.proider.com.5060 >
my.kamailio.com.5068: UDP, length 1882</div>
<div>E..... ./...6...</div>
<div>........b..SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP
my.kamailio.com:5068;rport=5068;received=<a href="http://my.kamailio.com" target="_blank">my.kamailio.com</a>;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2</div>
<div>Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600</div>
<div>Record-Route: <sip:<a href="http://my.proider.com" target="_blank">my.proider.com</a>;lr=on;ftag=as5872f19e></div>
<div>Record-Route: <a><sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on></a></div>
<div>From: <<a href="mailto:sip%3ATrunkNum@my.provider.com" target="_blank">sip:TrunkNum@my.provider.com</a>>;tag=as5872f19e</div>
<div>To: <<a href="http://sip:12345678900@my.provider.com:5068" target="_blank">sip:12345678900@my.provider.com:5068</a>>;tag=5rF0FNamQ99gH</div>
<div>Call-ID: <a href="mailto:42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600" target="_blank">42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600</a></div>
<div>CSeq: 102 INVITE</div>
<div>Contact: <a><sip:12345678900@externail.number.end.ip:5060;transport=udp></a></div>
<div>User-Agent: provider agent</div>
<div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
MESSAGE, INFO, UPDATE, REFER, NOTIFY</div>
<div>Supported: timer, precondition, path,
replaces</div>
<div>Allow-Events: talk, hold, conference, refer</div>
<div>Content-Type: application/sdp</div>
<div>Content-Disposition: session</div>
<div>Content-Length: 746</div>
<div>X-provider agentOutboundGateway: <a href="mailto:sip%3A5574012345678900@62.93.147.149" target="_blank">sip:5574012345678900@62.93.147.149</a></div>
<div>X-provider agentOutboundCarrierID:
23705946361020</div>
<div>X-provider agentCarrierRate: 0.20180</div>
<div>X-provider agentCloudRate: 0.00300</div>
<div>Remote-Party-ID: "Outbound Call" <<a href="mailto:sip%3A5060@my.provider.com" target="_blank">sip:5060@my.provider.com</a>>;party=calling;privacy=off;screen=no</div>
<div><br>
</div>
<div>v=0</div>
<div>o=FreeSWITCH 1409844386 1409844387 IN IP4
externail.number.end.ip</div>
<div>s=FreeSWITCH</div>
<div>c=IN IP4 externail.number.end.ip</div>
<div>t=0 0</div>
<div>a=msid-semantic: WMS
hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP</div>
<div>m=audio 23216 RTP/AVP 0 101</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div>
<div>a=silenceSupp:off - - - -</div>
<div>a=ptime:20</div>
<div>a=ssrc:2990874569 cname:pRs5xP</div>
<div><br>
</div>
<div> </div>
<div><br>
</div>
<div> </div>
<div><br>
</div>
<div>IP my.kamailio.com.5068 >
my.proider.com.5060: UDP, length 596</div>
<div><a href="mailto:E..p.@..@.K%5C" target="_blank">E..p.@..@.K\</a></div>
<div>...6........\`.ACK <a>sip:12345678900@externail.number.end.ip:5060;transport=udp</a>
SIP/2.0</div>
<div>Via: SIP/2.0/UDP
my.kamailio.com:5068;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0</div>
<div>Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600</div>
<div>Route: <sip:<a href="http://my.proider.com" target="_blank">my.proider.com</a>;lr=on;ftag=as5872f19e></div>
<div>Max-Forwards: 70</div>
<div>From: <<a href="mailto:sip%3ATrunkNum@sip.callsion.com" target="_blank">sip:TrunkNum@sip.callsion.com</a>>;tag=as5872f19e</div>
<div>To: <<a href="http://sip:12345678900@my.provider.com:5068" target="_blank">sip:12345678900@my.provider.com:5068</a>>;tag=5rF0FNamQ99gH</div>
<div>Call-ID: <a href="mailto:42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600" target="_blank">42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600</a></div>
<div>CSeq: 102 ACK</div>
<div>User-Agent: Asterisk PBX 12.5.0</div>
<div>Content-Length: 0</div>
<div><br>
</div>
<div> </div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div> </div>
<div><br>
</div>
<div> </div>
<div><br>
</div>
<div>IP my.kamailio.com.5068 >
my.proider.com.5060: UDP, length 617</div>
<div><a href="mailto:E....D..@.KC" target="_blank">E....D..@.KC</a></div>
<div>...6........q.ZBYE <a>sip:12345678900@externail.number.end.ip:5060;transport=udp</a>
SIP/2.0</div>
<div>Via: SIP/2.0/UDP
my.kamailio.com:5068;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0</div>
<div>Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600</div>
<div>Route: <sip:<a href="http://my.proider.com" target="_blank">my.proider.com</a>;lr=on;ftag=as5872f19e></div>
<div>Max-Forwards: 70</div>
<div>From: <<a href="mailto:sip%3ATrunkNum@sip.callsion.com" target="_blank">sip:TrunkNum@sip.callsion.com</a>>;tag=as5872f19e</div>
<div>To: <<a href="http://sip:12345678900@my.provider.com:5068" target="_blank">sip:12345678900@my.provider.com:5068</a>>;tag=5rF0FNamQ99gH</div>
<div>Call-ID: <a href="mailto:42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600" target="_blank">42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600</a></div>
<div>CSeq: 103 BYE</div>
<div>User-Agent: Asterisk PBX 12.5.0</div>
<div>X-Asterisk-HangupCause: Normal Clearing</div>
<div>X-Asterisk-HangupCauseCode: 16</div>
<div>Content-Length: 0</div>
<div><br>
</div>
<div><br>
</div>
<div>IP my.proider.com.5060 >
my.kamailio.com.5068: UDP, length 568</div>
<div>E..T....-.676...</div>
<div>........@..SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP
my.kamailio.com:5068;received=<a href="http://my.kamailio.com" target="_blank">my.kamailio.com</a>;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0</div>
<div>Via: SIP/2.0/UDP
my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600</div>
<div>From: <<a href="mailto:sip%3ATrunkNum@sip.callsion.com" target="_blank">sip:TrunkNum@sip.callsion.com</a>>;tag=as5872f19e</div>
<div>To: <<a href="http://sip:12345678900@my.provider.com:5068" target="_blank">sip:12345678900@my.provider.com:5068</a>>;tag=5rF0FNamQ99gH</div>
<div>Call-ID: <a href="mailto:42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600" target="_blank">42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600</a></div>
<div>CSeq: 103 BYE</div>
<div>User-Agent: provider agent</div>
<div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
MESSAGE, INFO, UPDATE, REFER, NOTIFY</div>
<div>Supported: timer, precondition, path,
replaces</div>
<div>Content-Length: 0<br>
<br>
<br>
<br>
<br>
Thanks for help.</div>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
</div>
</div>
<pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><span><font color="#888888">
</font></span></pre>
<span><font color="#888888"> </font></span></blockquote>
<span><font color="#888888"> <br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Next Kamailio Advanced Trainings 2014 - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
Sep 22-25, Berlin, Germany</pre>
</font></span></div>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</blockquote>
<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Next Kamailio Advanced Trainings 2014 - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
Sep 22-25, Berlin, Germany</pre>
</div></div></div>
</blockquote></div><br></div></div>