1 Feb
2017
1 Feb
'17
7:28 a.m.
Hi all,
Please count me and one other in for the meal on Saturday.
See you in Brussels!
Hugh Waite
On 30 Jan 2017 11:00 a.m., <sr-users-request(a)lists.sip-router.org> wrote:
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Today's Topics:
1. Diversion Header - relay messed up (Roman Dissauer)
2. Re: dlg_set_timeout bye to websockets client (Switch168)
3. Presence Module do not load db_url (Vu. Minh Cao)
4. Re: FOSDEM 2017 (Giacomo Vacca)
5. Re: Replace old registration record with new one while
keeping single AOR per contact (Federico Cabiddu)
6. Re: Replace old registration record with new one while
keeping single AOR per contact (Aqs Younas)
7. Re: mhomed opened sockets (Diego Nadares)
8. Re: [sr-dev] FOSDEM 2017 (Daniel-Constantin Mierla)
9. Re: FOSDEM 2017 (Ivan Todorov)
10. Re: [Kamailio-Business] [sr-dev] FOSDEM 2017 (Alexandr Dubovikov)
----------------------------------------------------------------------
Message: 1
Date: Sun, 29 Jan 2017 18:42:23 +0100
From: Roman Dissauer <roman(a)dissauer.net>
To: sr-users(a)lists.sip-router.org
Subject: [SR-Users] Diversion Header - relay messed up
Message-ID: <DB3FC45E-ACCE-426E-B662-031909F6BA40(a)dissauer.net>
Content-Type: text/plain; charset=utf-8
When I get an INVITE with Diversion Header the Request is forwarded without
Diversion Header and the Request User is taken from Diversion User.
Problem is that on the Destination Host I cannot get original Request User
what is the intended destination!
Is this intended behaviour? How can I change this behaviour?
I already tried to delete the Diversion Header on request_route but this
didn’t change the behaviour.
Thanks in andvance for your help!
Roman
Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as
Carrier IP
Incoming INVITE:
INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0.
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843.
From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb.
To: <sip:436761234567@x.x.x.x>.
Call-ID: 906143813_44460603(a)y.y.y.y.
CSeq: 28100 INVITE.
Max-Forwards: 19.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,
NOTIFY,PRACK,UPDATE,OPTIONS.
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed.
Contact: <sip:+43987654321@y.y.y.y:5060>.
P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>.
Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no;
reason=unconditional; counter=1.
Supported: timer,100rel.
Session-Expires: 1800.
Min-SE: 90.
Content-Length: 260.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
...
Relayed INVITE:
INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0.
Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>.
Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>.
Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582.
4fc0216dbecafde29127db502993222c.0.
Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch=z9hG4bK02B7871714d9345f843.
From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb.
To: <sip:436761234567@x.x.x.x>.
Call-ID: 906143813_44460603(a)y.y.y.y.
CSeq: 28100 INVITE.
Max-Forwards: 18.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,
NOTIFY,PRACK,UPDATE,OPTIONS.
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed.
Contact: <sip:+43987654321@y.y.y.y:5060>.
P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>.
Supported: timer,100rel.
Session-Expires: 1800.
Min-SE: 90.
Content-Length: 274.
Content-Disposition: session; handling=required.
...
------------------------------
Message: 2
Date: Sun, 29 Jan 2017 18:21:37 +0000
From: Switch168 <team(a)switch168.com>
To: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.sip-router.org>rg>, miconda(a)gmail.com
Subject: Re: [SR-Users] dlg_set_timeout bye to websockets client
Message-ID:
<CAJ4KJRjeZT++vYWkfnY10WnkWztpVtpSV700TW0ebh7z+jRAEg(a)mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
Hi Daniel,
For anyone else who might run into this issue I found
https://github.com/kamailio/kamailio/issues/85 to be related to mine. And
its a good a starting point.
Cheers
Andrew
On Fri, Jan 27, 2017 at 10:47 AM Switch168 <team(a)switch168.com> wrote:
HI Daniel,
modparam("dialog", "dlg_flag", 4) modparam("dialog",
"send_bye", 1)
modparam("dialog", "timeout_noreset", 1)
So regular bye's by useragent i can succesfully relay to this
<sip:6gjlmali@ec2c66jsa0ei.invalid;transport=ws>
style of uri by using this snippet below
# Handle requests within SIP dialogs
route[WITHINDLG] {
if(is_method("BYE")) {
xlog("DEBUG: Received BYE");
route(NATDETECT);
loose_route();
dlg_manage();
t_check_trans();
handle_ruri_alias();
rtpengine_delete();
route(RELAY);
exit;
}
...
But the byes that are sent out by dlg_set_timeout("$var(timer)") gives out
error cannot resolve
the random.uri and I don't know how to fix the bye before it get sent out.
Thanks
Andrew
On Fri, Jan 27, 2017 at 12:15 AM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
Hello,
isn't the dialog module setting the right value there? What function are
you using to update the contact?
Cheers,
Daniel
On 27/01/2017 06:01, Andrew Tan wrote:
Hello,
Just wondering if it's possible to intercept the bye message that
dlg_set_timeout sends out to do some nat_helper function to fix it so the
bye can be sent to random.invalid uris.
I know there is the edge proxy outbound module but wondering if I can do
it without.
Regular bye between 2 clients i was able intercept and fix with nat helper
but I dont know how to fix the byes that is sent out from dlg_set_timeout
function.
Thanks in advance!
Andrew
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--
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