similar to: Fast AGI library/support for C & C++

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

Hi list, I try to install asterisk on vps server , but fails when I want to install dahdi [root at shark dahdi-linux-2.6.3-rc1]# make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' You do not appear to have the sources for

Hi I have a problem w/ voicemail, the subject message is corruption when used voicemail variables, e.g. : voicemail.conf emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR} Return: Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?= Expected: Subject: 1504|12|"Teste - Rafael" <1570>|16 Thank's Att, *Rafael dos Santos Saraiva* Tel: (51)

Starting with 3.7 rc1, my workstation seems to loose ram. Up until (and including) 3.6, used-(buffers+cached) was roughly the same as sum(rss) (taking shared into account). Now there is an approx 6G gap. When the box first starts, it is clearly less swappy than with <= 3.6; I can''t tell whether that is related. The reduced swappiness persists. It seems to get worse when I update

[The is an announcement to the developers list only, please give the new version a try overnight and I will announce it to the wider community tomorrow.] This is to announce the release of openssh-2.1.0, the first stable release of portable OpenSSH to incorporate support for the SSH2 protocol. The SSH2 protocol offers a number of advantages over the SSH1 protocol including standards compliance

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

e deleflie <edeleflie at gmail.com> ... > ok, I do realise that the extending the maximum channel count may be > difficult ... there's gotta be a way to do it though. Perhaps a comparison of the FLAC structure with that of Vorbis will help. Within a FLAC stream the audio is split into blocks which are grouped (interleaved). But within each block the eight channels are chained

Hi List Apologies if this isn''t the correct place for this query (I''ve tried a search of the mail archives but not had much joy). I''m running R (2.14.0) on a Mac (OSX v 10.5.8, 2.66GHz, 4GB memory) and am having a few performance issues with reading data in from a Postres database (using RPostgreSQL). My query / code are as below # -----------------------------

>>>>> "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

With the ATI bug I was hitting earlier fixed, only my btrfs partition continues to show high cpu usage for some operations. Rsync, git pull, git checkout and svn up are typicall operations which trigger the high cpu usage. As an example, this perf report is from using git checkout to change to a new branch; the change needed to checkout 208 files out of about 1600 total files. du(1) reports

Tried to do a plot just now and got an error: Error in text.default(2, 6, main, cex = cex) : X11 font -adobe-helvetica-%s-%s-*-*-%d-*-*-*-*-*-*-*, face 1 at size 16 could not be loaded This is reproducible (in my current environment) via: par(mfrow=c(2,2)) plot(1:10) text(2,6,"Oh, hell!",cex=2) This is of course not the real context in which the error originally manifested

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

Hi, I hope you might give me some hints on how to find where my configuration is wrong, I am new to Asterisk and do not know, how to find the problem. Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the same maschine: Hylafax fax server. I want hylafax to use t38modem (a virtual T.38 modem) for sending faxes. t38modem schould connect to asterisk on the same host. If hylafax

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

>>>>> "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

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

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>

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 =>

Hi I would like the opinion of you and if anyone has a similar scenario. I have a project for installation of a Asterisk server in a client with about 400 extensions. My question is whether this scenario carry an Asterisk virtualized. Will be used only extensions and trunks sip sip, 1 queue with 2 agents, without call recording. It is best to use XEN or VMware? Which best version of Asterisk for