similar to: Weird SIP stuff

On Fri, Apr 24, 2015 at 9:59 AM, G?k?en Eraslan <gokcen.eraslan at gmail.com> wrote: [...] > > But "Watch" only notifies when there are new pull requests and issues, > which doesn't make sense for the r-source repository. Following Github Atom > feed[1] sounds better, however the feed only provides commit messages not > the diffs. > Right, sorry, I

>>>>> "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.

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

Hi all, There's a good chance that the people who most need to see this won't see it, but here goes anyway. Google is currently dropping a _lot_ of the mail we attempt to deliver to lists.fd.o subscribers. The immediate cause is sending on mail from domains with SPF/DKIM/DMARC policies which explicitly specify that lists.fd.o cannot relay mail on their behalf. Every time we do that, not

>>>>> "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

On 2017-04-18 08:31 PM, Victor Villarreal wrote: > Maybe excecuting the following command at Asterisk console, will help you: > > asterisk> voicemail show users > > And you will get a list of all mailbox configured in your system. Search > for the user with problems. VoiceMail stocktrans2 Angelica Douglas 12 Definitely there. In fact, I generate all

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

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

Hi, This is an early warning: OpenSSH will drop tcpwrappers in the next release. sshd_config has supported the Match keyword for a long time and it is possible to express more useful conditions (e.g. matching by user and address) than tcpwrappers allowed. Removing it reduces the amount of code in the 'hot' pre-authentication path in sshd and rids us of a dependency. -d

https://bugzilla.mindrot.org/show_bug.cgi?id=2695 Bug ID: 2695 Summary: inconsistent outout of "ssh.add -l" between ed25519 and rsa keys Product: Portable OpenSSH Version: 7.3p1 Hardware: Other OS: Linux Status: NEW Severity: minor Priority: P5 Component:

Dear All, Is there anyone who is having FastAGI support for C & C++? We do have FastAGI working for the JAVA and rest of the language / script. But I am unable to find FastAGI for C/C++. Please let us know how to write FastAGI using C/C++. Thanks in Advance, Kashyap -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an error sending to a specific fax number. It may be more than one but this is the one I investigated. It seems the delay for the SIP negotiation in T.38 was initiated after 6 seconds, however, our system sent the BYE after only 4 seconds, possibly cutting the call before all the communication necessary for the negotiation was completed. Here is the trace from our provider showing their

>>>>> "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

Joshua So once a transport is pulled from the transports table in realtime during asterisk startup it can't get any updates? Can a new transport be added to the table and the associated endpoints be updated to use the new transport, or are transport types only read at startup across the board? Thanks Bryant ---------------------------------------- From: "Joshua

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

if the dns has _sip._tcp.my.dom. SRV 0 0 5060 asterisk.dom.ain. _sip._udp.my.dom. SRV 0 0 5060 asterisk.dom.ain. where asterisk.dom.ain. is an A RR of the asterisk pbx. how does a call to sip://user@my.dom come in to asterisk so i can route it? do i just put in sip.conf [username] context = from-url-username and extensions.com [from-url-username] exten =>

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

Hello: Thanks for the input so far. Heres the issue-- This is a production environment-- where many people "touch" the files. ie-- The audio engineer is a freelancer who wants to master the files at the highest quality TO HIS EAR and experience-- He knows NADA, Not a thing about SOX-- but is a ProTools GURU. The SOX resampled files work on our asterisk box-- but I gotta put someone

Hi, we have a faxserver with Asterisk, IAXModem and Hylafax. Faxes come from a SIP trunk to Asterisk, then are forwarded throught 5 IAXModems managed with Hylafax. Hylafax users can also send faxes to these modems and Asterisk send them throught the SIP trunk. We also have a dedicated modem used only for sending faxes coming from an Hylafax dedicated user. Sometimes Hylafax reports that a modem