Re: [SR-Users] Installing rtpengine from RPM on CentOS

018 13:37 To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Installing rtpengine from RPM on CentOS This is packaged in main kamailio rpm package https://github.com/kamailio/kamailio/blob/master/pkg/kamailio/obs/kamailio.spec#L1453-L1454 чт, 1 нояб. 2018 г. в 14:33, Pan Chris

[SR-Users] Installing rtpengine from RPM on CentOS

Hello! According to https://github.com/sipwise/rtpengine/blob/master/el/README.el.md , we should be able to install rtpengine from a CentOS repository using yum. We're unable to find the packages. Any suggestions? Pan B. Christensen Developer Phonect AS Mail: pan.christen...@phonect.no

Re: [SR-Users] Missing documentation ims_usrloc_scscf module

; From: Henning Westerholt > Sent: tirsdag 16. oktober 2018 20:27 > To: sr-users@lists.kamailio.org > Cc: Carsten Bock ; Pan Christensen > > Subject: *** SPAM *** Re: [SR-Users] Missing documentation > ims_usrloc_scscf module > > Am Dienstag, 16. Oktober 2018, 13:31:08 CEST

Re: [SR-Users] Missing documentation

oter] From: sr-users On Behalf Of Pan Christensen Sent: tirsdag 16. oktober 2018 09:43 To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Missing documentation Also missing from devel: Not Found The requested URL /docs/modules/devel/modules/ims_usrloc_scscf.html was not found on

Re: [SR-Users] Missing documentation

...@phonect.no<mailto:pan.christen...@phonect.no> + 47 41 88 88 00 [mail_footer] From: sr-users On Behalf Of Pan Christensen Sent: tirsdag 16. oktober 2018 09:14 To: Kamailio (SER) - Users Mailing List Subject: [SR-Users] Missing documentation Hello. The documentation for the ims_usrloc_scscf

[SR-Users] Missing documentation

Hello. The documentation for the ims_usrloc_scscf module seems to be missing: Not Found The requested URL /docs/modules/stable/modules/ims_usrloc_scscf.html was not found on this server. Apache/2.4.10 (Debian) Server at www.kamailio.org Port 443 Pan B. Christe

Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

Or maybe FreeSwitch is redundant if you use rtpengine… With kind regards Pan B. Christensen Developer Phonect AS From: sr-users On Behalf Of Emanuel Gianico Sent: fredag 15. juni 2018 13:29 To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio + FreeSwitch + WebRTC I'm goin

Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

> On Fri, Jun 15, 2018 at 08:21:56AM +0000, Pan Christensen wrote: > > > I based my > > > test on https://github.com/havfo/WEBRTC-to- > > > SIP/blob/master/etc/kamailio/kamailio.cfg > > > but I blamed my failure on an old rtpengine :) > > > >

Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

> I based my > test on https://github.com/havfo/WEBRTC-to- > SIP/blob/master/etc/kamailio/kamailio.cfg > but I blamed my failure on an old rtpengine :) You need to add 'SDES-off' to the rtpengine_manage strings for calls going to WebRTC. Most browsers don't support fallback to SDES (anymore) an

Re: [SR-Users] outbound flow tokens and kamailio restart

Bounce. Med vennlig hilsen Pan B. Christensen From: sr-users On Behalf Of Pan Christensen Sent: onsdag 13. juni 2018 09:19 To: Kamailio (SER) - Users Mailing List Subject: [SR-Users] outbound flow tokens and kamailio restart Hello all. I have created a WebRTC to SIP gateway. I implemented

Re: [SR-Users] Asterisk inviting Kamailio

This question probably belongs in the asterisk forum, but here's a quick answer: It's most likely a reINVITE in order to modify the existing session. Compare it to the original and see if anything has changed (like the codecs in the SDP). Med vennlig hilsen Pan B. Christensen Utvikler Phonect AS

[SR-Users] outbound flow tokens and kamailio restart

Hello all. I have created a WebRTC to SIP gateway. I implemented it using the outbound module. If I restart Kamailio during a call, subsequent messages fail to be routed. {1 322 BYE 218565972_26748892@x.x.x.x} INFO: outbound [outbound_mod.c:261]: decode_flow_token(): flow-token failed validati

Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio

Dear Steve, Would you mind sharing your findings and solution with the list? With kind regards Pan B. Christensen Developer Phonect AS  > -Original Message- > From: sr-users On Behalf Of Wilkins, > Steve > Sent: mandag 11. juni 2018 12:30 > To: Kamailio (SER) - Users Mailing List > Sub

Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

VP8 Is this correct? Then I think one of the issues is that there is no fmtp line in the VP8. The only codecs that have fmtp lines is for the H264 codecs. Thank you, -Steve From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan Christensen Sent: Friday, June 8, 2

Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

ing VP8 Is this correct? Then I think one of the issues is that there is no fmtp line in the VP8. The only codecs that have fmtp lines is for the H264 codecs. Thank you, -Steve From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan Christensen Sent: Friday, June 8, 2018

Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

(VP8) calls a Soft-Phone (H264). What is strange is that if it is the other way around and the Soft-Phone calls the WebRTC client, it works. Thank you From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan Christensen Sent: Friday, June 8, 2018 9:00 AM To: Kamailio (

Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello Steve. Does Asterisk negotiate different codecs with each client? If so, it needs to transcode, which I believe is currently not supported for video. What does Asterisk send back to device A? With kind regards Pan B. Christensen Developer Phonect AS From: sr-users On Behalf Of Wilkins,

[SR-Users] (N)DB_FIREBASE module?

Greetings all. We are currently developing web and mobile apps using firebase: https://firebase.google.com/products/realtime-database/ . We have been discussing the possibility of Kamailio querying this database. I assume that it will be possible to query it using HTTP and JSON modules, b

Re: [SR-Users] configuring websockts in kamilio

Hello. The js client cannot know this information, and hence cannot send it. What you’re seeing is normal for SIP over websockets. It should just work. What issues are you experiencing? With kind regards Pan B. Christensen Developer Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway

Re: [SR-Users] General Kamailio with Asterisk (or another PBX) Opinion

Hello, This answer is somewhat based on previous questions that you have asked. > Who forwards Registrations to Asterisk or PBX, and who lets Kamailio > maintain Registrations? As Alex said, it depends on what you are trying to accomplish. In addition, I would say that it depends on the call sc

Re: [SR-Users] INFO: stun_parse_header: incomplete header of STUN message

Dear Daniel. Thanks for your quick reply and fix. The proper solution for us is to make the backend stop sending keepalives as they are not needed in this case. I've asked our supplier about this as I couldn't find an option for it. I'm assuming that the backend incorrectly detects NAT and the

[SR-Users] INFO: stun_parse_header: incomplete header of STUN message

Hello! After proxying a REGISTER from a WebRTC client to our SIP backend, the backend starts sending keepalives. Every 30 seconds I get a log entry in Kamailio saying: "INFO: stun [kam_stun.c:169]: stun_parse_header(): INFO: stun_parse_header: incomplete header of STUN message". Here's one su

Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

Hello Steve. What are you trying to achieve? The call could go from client A to Kamailio to client B. No need to involve Asterisk. If you need PBX functionality, the INVITE needs to be routed to Asterisk, which will most likely answer the call and then set up a new call to client B. As Asteris

Re: [SR-Users] Understanding port changing

Hello! This is probably caused by SIP ALG in the NAT router. Compare what the client sends to what you receive on the server. With kind regards Pan B. Christensen Developer Phonect AS > -Original Message- > From: sr-users On Behalf Of Social > Boh > Sent: onsdag 9. mai 2018 18:32 > To:

Re: [SR-Users] WebRTC to SIP gateway

er suggestions? With kind regards Pan B. Christensen Developer Phonect AS From: sr-users On Behalf Of Pan Christensen Sent: onsdag 9. mai 2018 15:09 To: sr-users@lists.kamailio.org Subject: [SR-Users] WebRTC to SIP gateway Hello! It's been several years since I've used Kamailio. My cur

[SR-Users] WebRTC to SIP gateway

Hello! It's been several years since I've used Kamailio. My current employer wants to implement WebRTC, which is currently not supported in our SIP backend, and asked if I could set up a Kamailio server as a gateway. I've been able to make calls in all directions between SIP and WebRTC clients