similar to: RFC 5922 (TLS Certificates) and Asterisk

Hi, I have to connect 3 asterisk servers,each of them being TLS server for his clients and connected in both way in TLS with both others asterisk, each having hi own Common Name. Is this possible? I set up 2 asterik's , one server and the other client, this is OK. But I can't deal with certificats generated on both servers. I tried to put tlscertfile ans tlscafile in the peer

Hi there I have a pbx (v16.10) on AWS (Ubuntu 18.04) with Freepbx (14) that I am trying to set up the proxy reSIProcate on the same host as pbx. I can make it all work when the proxy is on a different host but when the proxy is on the same host asterisk sends the media address as 127.0.0.1 which the end user then happily sends media to 127.0.0.1 but it doesn’t get anywhere. Asterisk then

Hello, I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for either. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for OpenSIPS or Kamailio with the following features: (a) Support an arbitrarily large number of

I've now prepared a blog about my experience setting up Asterisk 11 with repro as a SIP proxy for WebSocket clients: http://danielpocock.com/using-resiprocate-to-connect-asterisk-webrtc In particular, the focus is on the use of packages because that makes it faster for more people to deploy identical working systems. To get the demo running for the WebSocket client, I really only needed

I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :)

I've just come across this issue: https://issues.asterisk.org/jira/browse/ASTERISK-17727 I am strongly in support of TLS and I believe this issue will be a stumbling block for more and more users - because more and more CAs are using the intermediate certificate chains For example, the free startssl.com certs are trusted by Android phones now. I have a UA running on my phone against a SIP

Greetings, As a developer and consultant who spends considerable time on projects involving the fusion of Asterisk and products derived from the SER ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have found that there is a great volume of interest in this topic on the mailing lists associated with all communities involved, but a comparative lack of focus that results in

One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using?

Good morning everybody. My question is simple. Is there a way to perform relay register with Asterisk ? More precisely, I want my clients regiter to a Proxy Registrar (OpenSIPS/Kamailio) through my Asterisk : REGISTER REGISTER Client ------------> Asterisk ---------------> OpenSIPS So Asterisk keep a list of registered clients and only allows them to

Hi All, Has anyone passed the tests using Asterisk: http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html I presume the same rules apply for scaling and possibly have OpenSIPS/Kamailio on the front? Thanks. -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com

Hello To cut down on the number of hackers trying to break into an Asterisk server, I'd like to simply move the SIP port from the standard UDP 5060 to something non-standard. Since this server must be able to receive INVITEs from any SIP UA (server or client), it appears that I must add an SRV record in the DNS so that they can locate the server and the port used to reach it. _sip._udp SRV

Hey John, In one installation I have, we use several monitoring tools (nagios based and custom scripts based) and we have the following: ; Reply OK to SIP:OPTIONS [public] exten => s,1,Wait(1) same => n,Hangup : For Nagios exten => nagios,1,Wait(1) same => n,Hangup NOTES: 1- We have context=public in sip.conf, if you have anything else, you must update the dialplan above

Hi all, I saw Matt Jordan's recent Kamailio world talk and was interested in the idea he proposed of stripping out authentication and registration from asterisk and letting Kamailio handle it. All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding registrations to asterisk. In order to do what Matt suggested would I be correct in assuming I would have to use the

Hi Dovid, We can change the SDP in Kamailio, but Asterisk will still send its RTP from its default address. The remote end is strict about accepting RTP from the specified source and won't accept it. Have you any suggestions to solve that problem? Thank you. On Fri, 30 Oct 2020 at 14:49, Dovid Bender <dovid at telecurve.com> wrote: > Why not use OpenSips/Kamailoo in between?

used Kamalio to "supplement" the features that Asterisk either doesn't provide or doesn't provide in as nice a form as the OP desired - can't really speak beyond this as I am not one of them. ------=_NextPart_000_010C_01CB6EAA.3AC2C610 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <html

On Thu, Jan 29, 2015 at 2:43 AM, Kirill Marchuk <62mkv at 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

I've got this setup in a test context. [test] exten => s,hint,SIP/7124 exten => s,1,NoOP(Options to $EXTEN) same => n,Hangup() exten => _x.,hint,SIP/7124 exten => _X.,1,NoOP(Options to $EXTEN) same => n,Hangup() exten => Anonymous,hint,SIP/7124 exten => Anonymous,1,NoOP(Options to $EXTEN) same => n,Hangup() I added hints to see if that would make a difference

branch xen-unstable xen branch xen-unstable job build-i386 test xen-build Tree: qemuu git://xenbits.xen.org/staging/qemu-upstream-unstable.git Tree: xen git://xenbits.xen.org/xen.git *** Found and reproduced problem changeset *** Bug is in tree: xen git://xenbits.xen.org/xen.git Bug introduced: ae763e4224304983a1cde2fbb3d6e0c4d60b2688 Bug not present:

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

Hi, I would need to bridge a SIP call with a multicast RTP channel. Both sides are receiving and transmitting RTP. Googling, I saw that an app_rtppage, which was in the SVN for a while and its not there anymore. It did, I think, only partly what I need (it sent from SIP to the mcast ... not the other way around), but it was a start. Any idea how to do this? I also could use