15 Oct
2018
15 Oct
'18
12:33 p.m.
Yes there was a 180 reply from the Callee.
2018/10/11 12:34:57.535510 65.xx.xx.161:64877 -> 65.xx.xx.167:5070
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
65.xx.xx.167:5070;branch=z9hG4bK98e3.3392254b0b2d98dfc66f12cb7fdba746.0
Via: SIP/2.0/UDP 65.xx.xx.172:5060;rport=5060;branch=z9hG4bK694382a1
Record-Route: <sip:65.xx.xx.167:5070;r2=on;lr=on;did=bbf.658>
Record-Route: <sip:65.xx.xx.167;r2=on;lr=on;did=bbf.658>
Contact: <sip:238@10.17.0.35:64877>
To: "John"<sip:238@65.xx.xx.161
:64877;rinstance=8a315091627cc10b>;tag=6467b07f
From: "Robert" <sip:226@mypbx.net>;tag=as0ecef1c4
Call-ID: 1e82197b42f0173b25e70759753d4210(a)mypbx.net
CSeq: 102 INVITE
User-Agent: Bria 4 release 4.8.1 stamp 84929
Allow-Events: talk, hold
Content-Length: 0
2018/10/11 12:34:57.535631 65.xx.xx.167:5060 -> 65.xx.xx.172:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 65.xx.xx.172:5060;rport=5060;branch=z9hG4bK694382a1
Record-Route: <sip:65.xx.xx.167:5070;r2=on;lr=on;did=bbf.658>
Record-Route: <sip:65.xx.xx.167;r2=on;lr=on;did=bbf.658>
Contact: <sip:238@10.17.0.35:64877;alias=65.xx.xx.161~64877~1>
To: "John"<sip:238@65.xx.xx.161
:64877;rinstance=8a315091627cc10b>;tag=6467b07f
From: "Robert" <sip:226@mypbx.net>;tag=as0ecef1c4
Call-ID: 1e82197b42f0173b25e70759753d4210(a)mypbx.net
CSeq: 102 INVITE
Allow-Events: talk, hold
Content-Length: 0
On Mon, Oct 15, 2018 at 12:15 PM Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
Hello,
indeed, I was misled by the Route headers in INVITE, which looked like
inside a dialog, but the parameter in To header is rinstance.
Is there any 18x response?
Cheers,
Daniel
On 15.10.18 16:00, Sergiu Pojoga wrote:
Hi again,
Hmm... I don't see a To-tag in the INVITE, neither there's a 200OK to
provide because the UPDATE was sent out prior to the callee answering the
call.
If there should be a Route header in the UPDATE, it would it indicate a
bug with Asterisk firing off the UPDATE without a pre-set Route dictated by
the Path?
If that's the case, I suppose my options are:
1. reach out to Asterisk to investigate and fix it (unrealistic)
2. store the Route header from the initial INVITE in a AVP and insert
it later if an UPDATE follows. Would that break anything up?
Any other constructive suggestions?
Thanks.
On Mon, Oct 15, 2018 at 2:34 AM Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
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>;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
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>;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
CSeq: 103 UPDATE
Content-Length: 0
Much obliged.
On Fri, Oct 12, 2018 at 9:38 AM Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
--
Daniel-Constantin Mierla -- www.asipto.comwww.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
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:
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: 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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Kamailio (SER) - Users Mailing
Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- www.asipto.comwww.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
--
Daniel-Constantin Mierla -- www.asipto.comwww.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