search for: siphones

"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know your iPhone." --0015174c3c60a73ef5046656ca27 Content-Type: text/html; charset=windows-1252 Content-Transfer-Encoding: quoted-printable

Hi, It worked finally with GSM Codec only enabled at client side.. Initially with G.711 (u-low) , G.711 (A-low) and GSM it didn't work. All enabled by setting [CLI] sip set debug on I saw asterisk having following logs.. -- Remotely bridging SIP/macbook-00000041 and SIP/tharindu-00000042 set_destination: Parsing <sip:tharindu at 192.168.1.3<sip%3Atharindu at 192.168.1.3> :

ode.google.com/p/siphon/wiki/FAQ_en</a>), I can read :<o:p></o:p></span></font></p> <h3><b><font size=3D4 face=3D"Times New Roman"><span style=3D'font-size:13.= 5pt'>&quot;Why Siphon doesn't allow to receive a call when it doesn't run<o:p></o:p></span= ></font></b></h3>

I checked the app store and haven't found anything promising, but I figured I'd ask here. Does anyone know of a SIP or IAX client for a non-jailbroken iPhone that will communicate directly with a machine running Asterisk? I know that there's at least one offering that seems like it's essentially a proxy (App runs on iPhone, iPhone talks to 3rd party server, 3rd party server talks

On Mon, Jan 16, 2017 at 12:31 PM, Sean Silva via llvm-dev <llvm-dev at lists.llvm.org> wrote: > Do we have any open projects on LLD? > > I know we usually try to avoid any big "projects" and mainly add/fix things > in response to user needs, but just wondering if somebody has any ideas. > I'm not particularly active in lld anymore, but the last big item I'd

Hi Napi, On Mon, 2006-11-06 at 12:17 +0800, Mohd-Hanafiah Abdullah wrote: > On Sun, 2006-11-05 at 19:06 -0800, Reid Spencer wrote: > ... > > As for the __main function, its a gcc library call required by the > > compiler for program startup. The details vary but the call is needed. > > Amongst other things it will probably initialize your C++ static > >

On Sun, 2006-11-05 at 19:06 -0800, Reid Spencer wrote: ... > As for the __main function, its a gcc library call required by the > compiler for program startup. The details vary but the call is needed. > Amongst other things it will probably initialize your C++ static > constructors. Hi Reid: I'm not using gcc for this purpose but another C compiler called AMPC. It compiles C

> On Jan 16, 2017, at 1:47 PM, Sean Silva <chisophugis at gmail.com> wrote: > > > > On Mon, Jan 16, 2017 at 1:25 PM, Davide Italiano <davide at freebsd.org <mailto:davide at freebsd.org>> wrote: > On Mon, Jan 16, 2017 at 12:31 PM, Sean Silva via llvm-dev > <llvm-dev at lists.llvm.org <mailto:llvm-dev at lists.llvm.org>> wrote: > > Do we

On Mon, Jan 16, 2017 at 2:31 PM, Davide Italiano <davide at freebsd.org> wrote: > On Mon, Jan 16, 2017 at 2:07 PM, Mehdi Amini <mehdi.amini at apple.com> > wrote: > > > > On Jan 16, 2017, at 1:47 PM, Sean Silva <chisophugis at gmail.com> wrote: > > > > > > > > On Mon, Jan 16, 2017 at 1:25 PM, Davide Italiano <davide at freebsd.org>

"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know= your iPhone." --_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0248cworksmailcwo_ Content-Type: text/html; charset="us-ascii"

Over the last two weeks, we have had at least two "incidents" where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further).

On Mon, Jan 16, 2017 at 3:34 PM, Sean Silva <chisophugis at gmail.com> wrote: > > > On Mon, Jan 16, 2017 at 3:32 PM, Sean Silva <chisophugis at gmail.com> wrote: > >> >> >> On Mon, Jan 16, 2017 at 2:31 PM, Davide Italiano <davide at freebsd.org> >> wrote: >> >>> On Mon, Jan 16, 2017 at 2:07 PM, Mehdi Amini <mehdi.amini at

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

Hello. I would need some help trying to setup Asterisk 1.6.2.9-2+squeeze1 on a Debian 6.0 system. I'd like to use the Debian packages, hence the "strange" version number? Since I'm new to Asterisk, I'm trying to follow "The Asterisk Book" at http://www.the-asterisk-book.com/unstable/minimale-telefonanlage.html and created a VERY basic sip.conf; see

Would it make sense for xray instrumentation be part of -fprofile-generate? PGO will affect inlining decisions etc for the optimized binary, but the collected traces during the instrumented build would still have quite a bit of useful information. -- Sean Silva On Mon, Jan 16, 2017 at 4:33 PM, Xinliang David Li <xinliangli at gmail.com> wrote: > Google GCC records profile data (dynamic

Hi Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091213/1f81f008/attachment.pgp

That’s a very good point, the X-ray data that would be useful for the linker should be collected from a "profile-use” build to be accurate. IIUC, that’d look like: build with profile-gen -> run benchmarks -> build with profile-use + Xray -> run benchmarks -> build with profil-use and feed linker with X-ray data for layout optimization. — Mehdi > On Jan 16, 2017, at 8:25

I appologize for the long, new-ish question, but after a few days of trying to work a solution by reading through the list archives and WIKI and coming up with what I thought would work, I think I'm just not getting a fine detail. I titled this thread "Setting up Security Groups" because I'm trying to set up some sip user groups with certain calling rights, e.g., one group of

I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and

Hello All - I've been poking around the past few weeks, trying to familiarize myself with all of this. I am new to Linux, VoIP and Asterisk -- to be complete. This is my first exposure to all of these technologies. I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge 2400 and the install went well. I can log in and poke around in Linux and I even configured the box to be