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@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)
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Message: 1 Date: Sun, 29 Jan 2017 18:42:23 +0100 From: Roman Dissauer roman@dissauer.net To: sr-users@lists.sip-router.org Subject: [SR-Users] Diversion Header - relay messed up Message-ID: DB3FC45E-ACCE-426E-B662-031909F6BA40@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@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@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. ...
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Message: 2 Date: Sun, 29 Jan 2017 18:21:37 +0000 From: Switch168 team@switch168.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org, miconda@gmail.com Subject: Re: [SR-Users] dlg_set_timeout bye to websockets client Message-ID: CAJ4KJRjeZT++vYWkfnY10WnkWztpVtpSV700TW0ebh7z+jRAEg@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@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@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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