similar to: SRV record for non-standard SIP port?

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

DISCLAIMER: This code is free (I am not charging you to use it), but you might have to pay royalty fees to the G.729 patent holders for using their algorithm. I finished this last Saturday and have had it on an Asterisk machine for 5 days without a crash, so I'm hoping that means it's safe to release into the real world. This code has also been released on the -dev list. As it is

I was somewhat concerned reading Mark's posting earlier today. Obviously, things are very bad in the US at the moment. Their Government even deported Cat Stevens the other day (check http://news.bbc.co.uk/1/hi/england/london/3686992.stm ). Clearly, given the fact that Digium contributes so much to Asterisk, they shouldn't be forced to risk their company's future by hosting these

Can asterisk use SRV record from DNS? In example - I've got: _sip._tcp.gda.pl. SRV 0 0 5060 welbot.task.gda.pl. _sip._udp.gda.pl. SRV 1 0 5060 welbot.task.gda.pl. And it works with sipd (from Columbia) and vocal. I see that with asterisk it does not want to work... :-/ -- pozdr. Pawe? Go?aszewski --------------------------------- worth to see:

Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: <Proxy_1_> _sip._udp.example.com </Proxy_1_> However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine.

Hi all, Lately, I've seen an increase in the number of attacks against my system from the so-called "Friendly Scanner." When one of these script kiddies targets my server, all I see for symptoms is a few of my trunks become lagged due to server load and a stream of messages on the console that resemble this: [Aug 2 20:27:50] == Using SIP VIDEO CoS mark 6 [Aug 2 20:27:50] ==

Hi all, is it possible that asterisk uses two proxies with SRV? The enddevices are registered on one of the two Proxies (Kamailio). The two proxies communicate with each other. And asterisk can choose one of this proxies with SRV. asterisk | \ | \ Proxy1 Proxy2 I have tries to solve this problem with two trunks for this proxies and Dial(... at proxytrunk) but on this way the

Let's say I have a third-party device acting as a sip<-->pstn gateway, a cluster of three asterisk servers, and a teensy bit of dns knowledge. Let's now say those asterisk servers are a1.company.com at 192.168.0.1, a2.company.com at 192.168.0.2, and a3.company.com at 192.168.0.3. 1. If I setup round-robin dns like so: asterisk.company.com. IN A 192.168.0.1 asterisk.company.com. IN

Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given call to tollfree.sip-happens.com ares.sip-happens.com was chosen

Sorry for messed up text. Here it goes again: I am learning to use the gsDesign package. I have a question about Pocock and OBF boundary. As far as I can understand, these 2 boundaries require equal spacing between interim analyses (maybe this is not correct?). But looks like I can still use gsDesign to run an analysis based on unequal spacing:? >

Hi, I am learning to use your gsDesign package!?I have a question about Pocock and OBF boundary. As far as Iunderstand, these 2 boundaries require equal spacing between interim analyses(maybe this is not correct?). But I can still use gsDesign to run an analysisbased on unequal spacing:?gsDesign(k=2,test.type=2,timing=c(0.75,1),alpha=0.05,sfu='Pocock')Symmetrictwo-sided group sequential

Still failed. The first secret is in your email program settings, to use Plain Text format (at least for emails you send to this mailing list). The second secret tool to use is the reprex package to let you verify that your code example will do on our computers what it is doing on your computer before you send it to us. That will also involve giving us some sample data or referencing some data

For those that were interested I have attached the kamailio.cfg which we have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the following yum packages: kamailio.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-auth-ephemeral.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-bdb.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms

> On 23 Sep 2017, at 01:32, array chip via R-help <r-help at r-project.org> wrote: > > Hi, > > I am learning to use your gsDesign package! I have a question about Pocock and OBF boundary. As far as Iunderstand, these 2 boundaries require equal spacing between interim analyses(maybe this is not correct?). But I can still use gsDesign to run an analysisbased on unequal

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

Can anyone recommend a good WebRTC phone to use with Asterisk? I do not mind if it is commercial or open source. Customers are starting to ask for web solutions and we need to start testing. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

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 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

Works for me.... Thanks, Hristo Benev -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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

I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration (MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be