15 Oct
2018
15 Oct
'18
2:34 a.m.
Hello,
that seems to be a re-INVITE (has To-tag). I would need at least the
initial INVITE and the 200ok, along with the UPDATE request.
If the UPDATE is after the re-INVITE, it is missing the Route header as
in the re-INVITE.
Cheers,
Daniel
On 12.10.18 16:53, Sergiu Pojoga wrote:
Hi Daniel,
Certainly, below find the initial INVITE and the subsequent UPDATE, as
received by Kamailio(a)65.xx.xx.167. If those aren't sufficient, let me
know and if it's ok with you, I'll send the full pcap in private.
The dilemma in my mind is whether the UPDATE should have a pre-set
Route header, similar to how the INVITE has.
2018/10/11 12:34:57.339306 65.xx.xx.172:5060 -> 65.xx.xx.167:5060
INVITE sip:238@65.xx.xx.161:64877;rinstance=8a315091627cc10b SIP/2.0
Via: SIP/2.0/UDP 65.xx.xx.172:5060;branch=z9hG4bK694382a1
Max-Forwards: 70
Route:
<sip:65.xx.xx.167;lr;received=sip:65.xx.xx.161:64877;r2=on>,<sip:xx.xx.xx.167:5070;lr;received=sip:65.xx.xx.161:64877;r2=on>
From: "Robert" <sip:226@mypbx.net
<mailto:sip%3A226@mypbx.net>>;tag=as0ecef1c4
To: <sip:238@65.xx.xx.161:64877;rinstance=8a315091627cc10b>
Contact: <sip:226@65.xx.xx.172:5060>
Call-ID: 1e82197b42f0173b25e70759753d4210(a)mypbx.net
<mailto:1e82197b42f0173b25e70759753d4210@mypbx.net>
CSeq: 102 INVITE
Supported: replaces,
timer, path
Content-Type: application/sdp
Content-Length: 386
2018/10/11 12:35:06.096457 65.xx.xx.172:5060 -> 65.xx.xx.167:5060
UPDATE sip:238@10.17.0.35:64877;alias=65.xx.xx.161~64877~1 SIP/2.0
Via: SIP/2.0/UDP 65.xx.xx.172:5060;branch=z9hG4bK34fab05c
Max-Forwards: 70
From: "Robert" <sip:226@mypbx.net
<mailto:sip%3A226@mypbx.net>>;tag=as0ecef1c4
To:
<sip:238@65.xx.xx.161:64877;rinstance=8a315091627cc10b>;tag=6467b07f
Contact: <sip:226@65.xx.xx.172:5060>
Call-ID: 1e82197b42f0173b25e70759753d4210(a)mypbx.net
<mailto:1e82197b42f0173b25e70759753d4210@mypbx.net>
CSeq: 103 UPDATE
Content-Length: 0
Much obliged.
On Fri, Oct 12, 2018 at 9:38 AM Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote:
Hello,
you hve to provide the sip traffic for this case, the screenshot
doesn't show the sip headers used for routing in this case,
therefore grab the sip traffic for all sip messages in such
scenarion, either ngrep output or pcap file, and send it over to
see if some headers are missing or not set properly.
Cheers,
Daniel
On 11.10.18 21:03, Sergiu Pojoga wrote:
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference -- www.kamailioworld.com
Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
Hi ppl,
I have this problem with call transfer, may be someone can help.
The phone to the far right is registered with the Registrar to
the far left using two PATH headers (trespassing two proxy ports,
5070 then 5060).
As you can see in the graph below, after receiving the UPDATE
request, Kamailio relays it further from port 5060, I expect it
to be from 5070 just like the dialog forming INVITE and the
CANCEL afterwards.
image.png
The UPDATE has a to-tag, but unlike the original INVITE - it has
no Route header!???
route[*WITHINDLG*] {
if (!has_totag()) return;
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
...
}
else if ( is_method("ACK") ) {
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
record_route();
}
route(RELAY);
exit;
}
if ( is_method("ACK") ) {
...
}
# handle UPDATE method for in-dialog requests
if (is_method("*UPDATE*")) {
route(DLGURI);
record_route();
route(RELAY);
}
}
Thanks in advance.
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--
Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com>
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Kamailio World Conference -- www.kamailioworld.com
<http://www.kamailioworld.com>
Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
<http://www.asipto.com>