sr-users December 2012

sr-users@lists.kamailio.org

71 participants
113 discussions

by Raj Roy Ghandhi

Hi, I want to do web (HTML5 + WebRTC) Sip client which can do the video conference with multiple users. Current release of SIPML <http://www.sipml5.org/> does 1 to 1 call. I have no idea of conference with many users. Is it the client that we need to modify to accept call and join the conference ? Do I need to send INVITE with extra parameters ? Please advice me. Best Regards, Roy.

11 years, 5 months

Question registering sip providers in kamailio and forward all?

by Thomas Hoellriegel

Hi all, I.m. a newbi in Kamailio. My problem: I.m. a blind user and i programmed a voicechat in Freeswitch. 3 have 2 sip providers. vodafone.de sipgate.at and freenet.de. The problem: i must register these proviers in kamailio. And forward all incoming calls to Freeswitch, and outgoing calls from freeswitch to kamailio. Dtmf modes are: auto rfc2833 and inband. Then i call: dorf(a)dorf.blindi.net i like to forward these calls to my freeswitch box. the kamailio ip ist: 217.172.180.108 and the freeswitchbox: 217.172.170.120 Can your help me please? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

11 years, 5 months

NAT Traversal issue

by Raj Roy Ghandhi

Hi, My Kamalio development version works very well with websocket and webrtc clients. But when I try to call the guy in remote area (he had connected to the same server with 3G dongle) no voice and video. Here is how I have set it up. 1. Kamailio 3.4 development version running on public IP 2. NAT Traversal is done with RTPProxy 1.2. 3. IP Phones work very well. (phones are behind NAT) 4. Web page with WebRTC works well in LAN behind the NAT But I try to call a account which in logged into same Kamailio server we do not hear voice nor media. I have attached the sip capture into 2 files 1. LAN webrtc client->LAN client web page call 2. LAN webrtc client -> 3G Dongle webrtc client Please help me out to figure this out. Best Regards, Roy.

11 years, 5 months

New developer: Richard Good

by Daniel-Constantin Mierla

Hello, I want to announce that a new developer got GIT write access to repository: Richard Good - he is among those developing the IMS extensions for Kamailio (http://code.google.com/p/smile-ims-extensions/), to be merged in the next days to the GIT master branch. His git commit id is: richard.good My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

11 years, 5 months

modules_s/timer review

by Daniel-Constantin Mierla

Hello, I was comparing modules_s/timer with modules_k/rtimer to spot the differences. modules_s/timer is working only with core timers (fast and slow timers), while the modules_k/rtimer can start own timer processes, or register a task to main core timer. Other differences ... modules_s/timer: - run on milisecond basis - option to enable/disable timer modules_k/rtimer - run either on second or micro-second basis - option to execute the many route blocks on same timer So, apart of enable/disable timer option, rtimer can do more than what timer module offer (mili-seconds can be run as 1000x micro-second). Enabling/disabling a timer can be workarounded from script, using a shared variable tested at begininning of rtimer route, like: route[RTIMER] { if($shv(rtimer)==0) return; ... } Enabling will mean setting $shv(rtimer)=1, either from other parts of the config or via MI/RPC commands. Now, both modules can be kept, no naming conflict, it is just a matter of code maintenance if we should keep timer module. Enabling/disabling timers can be added to rtimer in the future, but it is unlikely to happen before next major release (unless someone else does it). Is anyone here using modules_s/timer module aware of other differences comparing with rtimer? Looking for comments to keep or obsolete modules_s/timer module ... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

11 years, 5 months

Kamailio v3.3.3 Released

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

11 years, 5 months

Kamailio v3.3.2 Released

by Daniel-Constantin Mierla

11 years, 5 months

Planning next minor release - v3.3.3

by Daniel-Constantin Mierla

Hello, I am considering releasing next minor version from branch 3.3 at the beginning of next week, Monday or Tuesday (Dec 17 or 18). For outstanding issues, please use the tracker or send to sr-dev mailing list. Developers that pushed fixed in the master branch should backport them in 3.3 before the next week starts. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

11 years, 5 months

dlg_manage() and add_contact_alias() problem

by Pavel Miskov

Hi all, I want to end dialog (Kamailio sends BYE to both parties) that is lasting longer then some defined time. everything is working fine when using old fix_nated_contact function, but when using new add_contact_alias function kamailio sends BYE to private addresses. 1. "kamctl fifo dlg_list" output when using fix_nated_contact(): caller_contact:: sip:bob@PUBLIC_IP:50784 callee_contact:: sip:alice@PUBLIC_IP:64472 2. "kamctl fifo dlg_list" output when using add_contact_alias(): caller_contact:: sip:bob@10.2.5.206:10000 callee_contact:: sip:alice@10.2.5.205:20000 dlg_manag() doesn't store contact alias for caller and callee so Kamailio is sending BYE to private addresses. Am I doing something wrong or I can't use possibility of ending dialog with contact aliases? Thank you Pavel

11 years, 5 months

Support

by Murali Krish

Hi this is Murali krishnan i am final student of master degree in Dalhousie University, I am Working on my project, i m need of a sip server which can support H.264 codec video calls...I created an account in iptel sip account when i used the account it doesn't support video calls. If i use SER will this support video calls? -- *Regards Murali Krish*

11 years, 5 months

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sr-users December 2012