You should send a pcap file with all packets, from first incoming INVITE
to kamailio. It is important to have both sides of signaling from
kamailio point of view, from first packet in that call.
Cheers,
Daniel
On 28/08/14 15:11, Yuriy Gorlichenko wrote:
All packets (INVITE,ACK,BYE) that comes from Asterisk and sends to
Provider handled by Kamailio (changed tU, fU and td and from d). so I
write to PLIVO this question, but they still answer to me nothing...
As I see my trace there are no simple muistakes (such as wrong dst or
wrong contact header).
AboutAsteirsk and Kamailio I think As Daniel that Auth is not Asterisk
problem.
Furthermore Asterisk works with kamailio without registration on
kamailio: ip-based dialog.
So Daniel - If you will have some time to see my trace I will be happy.
Thanks for answers and help.
I will thinkabout problem to and waiting answer.
2014-08-28 16:57 GMT+04:00 Daniel-Constantin Mierla <mico...@gmail.com
<mailto:mico...@gmail.com>>:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko <ovoshl...@gmail.com
<mailto:ovoshl...@gmail.com>> wrote:
Hello. I try to provide call scheme:
internal client -> asterisk -> Kamailio -> provider -> external
endpoint call
when I make call I see this:
asterisk kamailio provider
invite --> invite -->
<-- 407
ACK -->
invite w/Auth -->
<-- 100 <-- 100
<-- 180 <-- 180
<-- 183 <-- 183
<-- 200 <-- 200
ACK --> ACK -->
My problem with last ACK, that I send to provider. Provider
ignores it, and sends me some OK packets. As resultI can notend
session ( answer to BYE 481 - transaction does not exists). I
think it is wrong ACK but can not undrtand where I do mistake.
Well, by letting the proxy handle authentication the INVITE
tranction i closed without Asterisk knowing about it. So the ACK
sent from the proxy and from Asterisk is for the same
transaction, which messes things up. Asterisk does not know
anything about the second invite. Letting the proxy handle
authentiction breaks the SIP protocol in bad ways and is
generally not a good solution.
You may want to send another response to asterisk when you get
the 407 so Asterisk retries and use the retry as a trigger for
the second INVITE and add auth to that.
While breaking the cseq incrementation for authentication
(mentioned in the readme of uac), the Asterisk seems to do ok
here, because the ACK is coming from asterisk, but it is not
accepted by the provider.
The provider (having a plivo platform, based on the responses) is
running kamailio 4.1.2 in front (looking at 100 trying).
Authentication from kamailio to another kamailio using uac module
should work fine, as kamailio doesn't act as end user UAC and
doesn't care much of cseq.
I didn't have time to look at the sip trace properly, but Asterisk
should have nothing to do with the problem here, unless I missed
something from the description.
Cheers,
Daniel
--
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 ::: Oct 15-17, San Francisco, USA
_______________________________________________
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://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
