Re: [asterisk-users] best kernel for Asterisk
Hello, On 19.04.17 09:57, marek cervenka wrote: > hi, > > what kernel version are you using for asterisk? > > are you satisfied with distro kernel (centos 6 2.6.32, centos 7 3.10, > ...) ? > > are you using newer kernels from elrepo.org? > > which kernel features are most critical for Asterisk performance pattern? > I prefer to work with the standard kernel from the distro (using debian) for security reasons, unless there is a team in the company with very good kernel tuning knowledge. Probably one can squeeze some more performances with a custom kernel build, but in long time that typically becomes a maintenance nightmare. If needed, you can instead aim for horizontal scaling by deploying a farm of Asterisk systems with a sip proxy load balancer in front of it (well, I could be a bit biased, because I do work mostly with the kamailio sip proxy). Anyhow, to cut it straight, standard distro kernel worked fine for the deployments I was involved in. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any valid up-to-date info about Kamailio-Asterisk integration ?
From Kamailio point of view, the tutorial referred here (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb) should be quite actual. As Matt said, we do have new features with more recent releases 4.1.x and 4.2.x but the relevant parts in the relation with Asterisk (authentication, registration, etc.) are more or less the same. If Asterisk preserved pretty much its old realtime mechanism and database structure, then should be straightforward to adjust in case of small changes. I hope to get a new tutorial that uses latest Kamailio and Asterisk 13 in the near future, targeting to use ARI instead of database for making the integration of the two applications. Cheers, Daniel On 29/01/15 16:52, Matthew Jordan wrote: On Thu, Jan 29, 2015 at 2:43 AM, Kirill Marchuk 62...@mail.ru wrote: Hi all Have recently watched Matt Jordan's session on Kamailio World 2014 On slides 26-29 of his presentation (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) he speaks about a (completely new, for me at least) approach to build scalable telephony systems, using N instances of Kamailio and N instances of Asterisk Are there any whitepapers, howtos, implementation experience reports, whatever, available, that would describe such an approach in details and help some not-so-advanced admins to at least understand if is it what they need, or not exactly, or not at all ? We are planning to look closer at Kamailio (or any other proxy, like OpenSip) as a way to do both load-balancing and failover solutions, so that refusal of any Asterisk instance should have minimal possible effect on the overall system availability. The best documentation out there - that I'm personally aware of - is Daniel's guide on integrating Kamailio and Asterisk: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb While there have been quite a few improvements made in Asterisk (and I imagine, Kamailio as well) since that was written, that guide would be a good starting point, regardless of the versions involved. A lot of questions howevere arise, like: what if one SIP user got REGISTERed at Server 1, and the other on Server 3, so how can they call one another ? There are many different ways of handling this. First, you have to ask yourself what you want Asterisk and Kamailio to do in your set up. Some sample questions: * Who acts as the registrar? * Who manages subscriptions? * Should each Asterisk server have a special purpose, or should they be treated as a generic pool of media servers? * Should Asterisk be involved in 'normal' calls (two-party, no media manipulation), or should it only be used when special services are needed? Your goal, in any scenario, should be to keep the Asterisk dialplan as simple as possible. That typically means not placing customer specific logic in the dialplan, but instead relying on func_odbc to pull customer specific information from a database. In later versions (such as Asterisk 13), you can remove much of the logic from the dialplan and use ARI to build custom media applications. But no, not a lot of this is written down yet. Also, outbound registrations can be done from one instance at a time, say it's done from Server1 for Trunk1, so how can users, that got authenticated at Server2, call thru that registration (Trunk1) ? If your Asterisk servers are sitting behind Kamailio, they should probably just be registering to their Kamailio instances. Again, if Kamailio is handling the registration, identification, and authentication, then you probably don't want Asterisk doing any of that. You would instead just have Asterisk trust that Kamailio is sending it the right calls, and have it handle them accordingly. Also, Kamailio itself has to be protected from failing, and probably even from overload... That's pretty standard stuff for Kamailio. Would be great to read something in-depth about that -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
On 01/08/14 10:56, Olli Heiskanen wrote: Hi, I got ahead with my setup, this post helped me much: http://forums.digium.com/viewtopic.php?f=1t=90167sid=66fdf8cc4be5d955ba584e989a23442f At least the avpf setting had to be removed from sip.conf and put in the realtime db table, defined per client. I left the encryption setting in sip.conf. I had some problems calling from SIP client to another, then had to define avpf=no for those clients. Personally I don't like to use different settings to different clients, is there a way around this? With this setup I can make calls between SIP clients but not ws clients. My client (now I use sip.js) fails to parse the sdp - including the apparently correct rtp profile UDP/TLS/RTP/SAVPF - and sends back 488, which makes the call fail. I'd like to hear opinions from you guys which would be the correct place to handle this? My setup has Asterisk Kamailio realtime integration, and I use dispatcher in Kamailio to route calls to Asterisk. Kamailio sounds like the logical place, but I'd rather find a way to not change the rtp profile along the way, at least until the clients can support that one. To understand properly, you don't want to use rtpenging for srtp(webrtc)-rtp(classic sip) gatewaying? If yes, maybe you can partition the users (classic-sip and webrtc-sip), then use two asterisk instances with routing via kamailio. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind NAT and Kamailio -- Internal IP in SDP and not externip
Hello, On 6/28/13 4:29 PM, Johan Wilfer wrote: Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the external. This is the setup: Teleco - Kamailio - Asterisk SIP -- 1.2.3.4 10.0.0.1 -- 10.0.0.2 externip=1.2.3.5 localnet=10.0.0.0/255.255.255.0 RTP 1.2.3.5 (NAT:ed to 10.0.0.2) On an incomming call from the teleco - to kamailio (public addr) - to asterisk in the private net. Asterisk responds with the following SDP: v=0 o=root 1889 1889 IN IP4 10.0.0.2 s=session c=IN IP4 10.0.0.2 t=0 0 m=audio 23344 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Asterisk seems to think that because the proxy is on the localnet, the media is too, so it doesn't use the externip as the RTP-ip. This is a incomming call and the RTP ip of the other leg is another public address. So the RTP-ip should the public address (externip). If I connect to the teleco directly from the pbx (bypassing kamailio) Asterisk correctly uses the externip as the rtp-ip in the SDP. I know this is an old and unsupported version of Asterisk, but any input on the topic is welcome. If this is supported in later versions we can maybe work around until we migrate later. what I did when I had similar scenario was to let asterisk completely behind NAT, using only the local IP. I used rtpproxy running on the same host as kamailio to bridge the rtp between external and internal networks. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial
Hello, I spent a bit of time to update my Kamailio-Asterisk realtime tutorial to latest stable versions in both sides. The tutorial is available at: - http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I tried to use default names for asterisk database tables, where the structure was not changed, and different names for those that are a bit customized, in order to make it easier to spot it is something special with it. For this one I had no much time for extensive testing, relying on lot of feedback that I got for the previous version (kamailio 3.3 and asterisk 10). I hope it will be useful for many here to get started with integration of the two applications, just reply to me for any feedback. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk realtime database structure
On 8/4/12 10:38 AM, virendra bhati wrote: best link for asterisk realtime is below one http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example On Fri, Aug 3, 2012 at 1:51 PM, Leandro Dardini ldard...@gmail.com mailto:ldard...@gmail.com wrote: If you check the contrib/realtime/mysql directory in the source tree, you'll find scripts for almost all the tables. Thank you all for the hints! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial
Hello, I released an update to my series of Kamailio and Asterisk Realtime Integration, using the latest stable versions of the two projects, respectively 3.3.1 and 10.7.0. You can find it at: * http://asipto.com/u/68 The tutorial focuses on how to use Asterisk's database structure to perform authentication in Kamailio SIP server, along with user location, nat traversal, instant messaging, presence, a.s.o., offloading processing from Asterisk. Asterisk will still handle all the calls, enabling rich telephony such as MoH, transcoding, ring back, IVR, etc. Reusing as much as possible the Asterisk database makes the architecture presented in the tutorial easy to be applied to existing installations, without losing management interfaces or other admin tools. Hope it is useful for many folks out there. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk realtime database structure
Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage I have several table structures from the Asterisk 1.6, I dug for them in the code or found on the web when I wrote the tutorial about integration with Kamailio 3.1 (http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb), but hopefully now it is an easy way to get the db structure. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ITSPA 2012 Award for Open Source VoIP Projects
Hello, ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry: * http://www.itspaawards.org.uk/ Open Source VoIP Projects won a special category this year, Members' Pick, for providing a real value to VoIP Industry. I had the chance to attend the event in London and I have been selected to pick up the award. I made a news on the website of the project I am mainly involved in (Kamailio) with more details: * http://www.kamailio.org/w/2012/03/itspa-awards-2012-open-source-voip-projects/ As you would expect, a complete voip platform usually involves several open source projects, for components such as load balancers, registrar, proxy, gateways or media servers, thus the decision of ITSPA for awarding to the group. It was rare when Asterisk was not mentioned as part of the VoIP systems in use by the ITSPA members I spoke to, no surprise! A significant part of the award is therefore (to be paint with) Asterisk logo. As another long time user of Asterisk project, I take the opportunity to send again my thanks to the people behind the project. If anyone is looking for more insights (for news, blogs, personal curiosity) about the event, just drop me an email! Cheers, Daniel -- Daniel-Constantin Mierla Co-Founder Kamailio SIP Server - http://www.kamailio.org Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 and Kamailio 3.1 realtime integration tutorial
Hello, I got the time to upgrade my tutorial about Asterisk and Kamailio realtime integration to latest stable release of Kamailio, version 3.1.0 (out on Oct 6, 2010). You can find the document at: * http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb Besides making it work for v3.1.x, the Kamailio config file has some new features included: * IP authentication - can be enabled via define WITH_IPAUTH * TLS support - can be enabled via define WITH_TLS - TLS to UDP translation and vice-versa is done automatically by Kamailio in case you configure Asterisk on UDP * detection of DoS attacks - can be enabled via define WITH_ANTIFLOOD - banning automatically traffic from attacker IP addresses for a specific time interval * restructuring of configuration file for better modularity and highlighting of functionalities such as registrar, location server, within-dialog request routing Hope it is useful for some people within this community. Next step, naturally, is to upgrade the tutorial for latest Asterisk, 1.8.0, just needs some time to get familiar with it. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] new way of asterisk and kamailio (openser) realtime integration
Hello, I put together a new tutorial about asterisk realtime integration with kamailio (openser). This time the database used is the one of asterisk, also call routing logic is controlled by asterisk, here is the link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Practically is an easier way to scale starting from existing asterisk installations. The other (old) version I wrote for long time, using kamailio database and asterisk just for media services, is available at: http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x Hope is useful for some of you! Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: new way of asterisk and kamailio (openser) realtime integration
On 5/17/10 1:02 PM, Randy R wrote: On Mon, May 17, 2010 at 12:36 PM, Fred Posnerf...@teamforrest.com wrote: Same problem here. ---fred On May 17, 2010, at 6:28 AM, Alexandru Oniciuc wrote: kb.asipto.com isn't reachable: DNS doesn't resolve the domain name. Alex I see the DNS resolving on our Virginia server but not here in Europe, oddly enough. TTL is one day. internet is more and more broken :-) -- dns last update on this zone was at least one week ago, issue was reported friday as well, but now seemed to work ok from several points of world ... very strange anyhow, why takes so long ... kb.asipto.com is aliased to www.asipto.com if you want to fix it quickly in local hosts file or use opendns.org dns servers, they obey ttl and update caches accordingly. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration
hello, still reports of non-updated dns caches in various sites of of the world, so I redirected an older subdomain to the page: http://ngs.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Sorry for any inconvenience on the list, Daniel On 5/17/10 2:08 PM, Hristo Benev wrote: Works for me Thanks, Hristo Benev -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexandru Oniciuc Sent: Monday, May 17, 2010 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration kb.asipto.com isn't reachable: DNS doesn't resolve the domain name. Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Daniel-Constantin Mierla Inviato: lunedì 17 maggio 2010 12:01 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] new way of asterisk and kamailio (openser) realtime integration Hello, I put together a new tutorial about asterisk realtime integration with kamailio (openser). This time the database used is the one of asterisk, also call routing logic is controlled by asterisk, here is the link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Practically is an easier way to scale starting from existing asterisk installations. The other (old) version I wrote for long time, using kamailio database and asterisk just for media services, is available at: http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x Hope is useful for some of you! Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] openser admin training session at VoN Fall Boston
Hello, apologizes if the email looks too off-topic... Last minute arrangements allowed to host one day of OpenSER Admin Training session within VoN Fall Boston, Nov 1, 2007, course that will cover openser and asterisk integration for basic media services. I believe the event could bring more value to people attending Digium Asterisk World co-located with VoN, being just next day. For more details about the course and registration (free of charge), see: http://www.openser.org/mos/view/OpenSER-Admin-Course---Boston-2007/ Thank you, Daniel -- Co-Founder OpenSER http://www.openser.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
