similar to: srtp/dtls when sip is clear over lo

>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: JBB> tcpenable=yes JBB> tlsenable=yes JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB> tlsdontverifyserver=yes JBB> tlscipher=ALL JBB> tlsclientmethod=tlsv1 You are missing the tls key. The config name is

Other things to consider: The transport config, which can be in [general] or in a peer's [] block. if you want tls-only, use transport=tls it also accepts tcp, udp or a comma-separated list. if given a list, it tries them in order If you need ast to register over tls, use something like this: register => tls://username:xxxxxx at sip-tls-proxy.example.org (copied from the

Does vorbis have any niches of technical superiority over opus? Or is compatibility with older hard- and software the only benfit? Put another way, is there any reason to prefer vorbis over opus for music on new sortware or platforms? -JimC -- James Cloos <cloos at jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

On Tue, March 3, 2015 13:37, James Cloos wrote: >>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: > > JBB> tcpenable=yes > JBB> tlsenable=yes > JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt > JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt > JBB> tlsdontverifyserver=yes > JBB>

>>>>> "AS" == A J Stiles <asterisk_list at earthshod.co.uk> writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. -JimC -- James Cloos <cloos at

>>>>> "DC" == D'Arcy Cain <darcy at VybeNetworks.com> writes: DC> Perhaps someone can explain what t38timeout is supposed to do A 'git grep t38timeout' on the src leads one to res/res_fax.c, where one case see that it is the number of miliseconds to permit for t38 negotiation to complete once it starts. Ie after both sides select t38, until they

>>>>> "JC" == Joshua Colp <jcolp at digium.com> writes: JC> I disagree that it makes it worthless for a large number of JC> users. It's only within the last few days that a few people have run JC> into this particular issue where they have a public IP address that is JC> changing a lot and PJSIP does not support changing it without a JC> restart.

>>>>> D'Arcy Cain <darcy at VybeNetworks.com> writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/sdp. -JimC -- James Cloos <cloos at jhcloos.com>

Hi List, I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I have installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws the following error: "chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP

The IANA has pre-allocated id 4 for ed25519. If waiting on the IANA were a reason to delay applying the SSHFP_KEY_ED25519 patch, it needn't be any longer. I've proposed un-reserving hash type 0 to be a "NULL hash", for those who'd rather publish the public key unhashed. Even if zero for unhashed fails to gain traction, I hope to see something allocated for that. But

Hello I get the following error when using our Let's Encrypt ssl certificate for webRTC calls : [Jun 2 14:29:28] == DTLS ECDH initialized (secp256r1), faster PFS enabled [Jun 2 14:29:28] ERROR[27360][C-00000ae5]: res_rtp_asterisk.c:1441 ast_rtp_dtls_set_configuration: Specified certificate file '/etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' for RTP instance

Dear Martin, Does the work on nchar mean that bugs #16090 and #16091 will be resolved [1,2]? Thanks, Mark [1] https://bugs.r-project.org/bugzilla3/show_bug.cgi?id=16090 [2] https://bugs.r-project.org/bugzilla3/show_bug.cgi?id=16091 On Sat, Apr 25, 2015 at 11:06 PM, James Cloos <cloos at jhcloos.com> wrote: > >>>>> "GC" == G?bor Cs?rdi <csardi.gabor at

It has been may years since I've done anything with a dahdi fxo; much has changed in the interim and I havne't found answers googling. The fxo hw is installed on the pots line in parallel to existing pots phones. My goal is to have a blf on the sip phone which lights whenever any of the devices on the pots line are off hook and which, when pressed, INVITEs the asterisk box such that it

When testing chacha20-poly1305, I noticed that aes-gcm is significantly faster than aes-ctr or aes-cbs with umac. Even on systems w/o aes-ni or other recent instruction set additions. And there seems to be consensus in the crypto community that AEAD ciphers are the way forward. As such, it promoting the AEAD ciphers to the head of the preference list looks like a good idea. That would mean

The progress metre in scp(1) breaks when the tty is too wide. This patch is the effortless fix: ########################################################################### :; diff -u openssh-2.1.1p2/scp.c openssh-2.1.1p2+jhc/scp.c --- openssh-2.1.1p2+jhc/scp.c Thu Jun 22 07:32:32 2000 +++ openssh-2.1.1p2/scp.c Sat Jul 1 22:15:36 2000 @@ -1176,8 +1176,9 @@ i = barlength *

Hi all ! I am using sipML live demo page (http://sipml5.org/call.htm?svn=224#) in order to test WebRTC setup on my Asterisk PBX. I am using latest SVN version of Asterisk 11 (Asterisk PBX SVN-branch-11-r417677) If I make calls from softphones (Zoiper, X-Lite), which do not support DTLS at all, I can hear the Echo Test sound. BUT when I call from browser (I've tried latest Mozilla Firefox

>>>>> "JC" == Joshua Colp <jcolp at digium.com> writes: JC> This stems from PJSIP not being dynamic with transports (it JC> doesn't like its environment changed to that degree while JC> in use). I'm afraid if your IP changes you'd have to restart JC> Asterisk when you are using PJSIP. Wow. I say this having voted for pjsip over the listed

For a sip-only application, what exactly is required to ensure that calls completed via followme are reinvited? Can it at all? The code after outbound = findmeexec(targs, chan) calls ast_bridge_ call(). I don't see anything there which can cause a reinvite, yes? When the same peer is used for both the incoming and outgoing legs, it is a bit of a waste to proxy the rtp. And even when the

Before coding this in C, I wanted to test the idea out in R. But I'm unsure if the theory is well-founded. I have a (user-supplied) black-box function which takes R^n -> R^3 and a defined domain for each of the input reals. I want to send some samples through the box to determine an approximation of the convex hull of the function's range. (I'll use the library from

Hi All, Currently i am integrating webRTC demo. I have issue using firefox,someone suggest me to enable DTLS for webRTC working in firefox using Asterisk. I am using Asterisk 11.9.0. https://groups.google.com/forum/#!searchin/doubango/bhavik/doubango/Mv9u0YkNb90/55VElJ1TdY8J Can any one tell me how to enable DTLS ? -- Thanks, Bhavik Patel -------------- next part -------------- An HTML