search for: 750ms

Does anyone know the default E&M Wink timings for Nortel DID ports? The default settings on Asterisk are: ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce: Debounce timing (default 600ms)

...emendously, and still be able to be performed quick enough for live discussion over the air between two communications devices. I have connected two communications devices via the Ethernet, streaming audio via UDP, and the only latency is the internet service provider and net itself, about 200ms - 750ms depending upon connection. But I want to find something that will give me a much much lower bandwidth requirement than "fully uncompressed audio". I wrote my own quick compressor that cuts the bandwidth in about half, but still, want to see "how low can I go". hence the current...

Hi, When I use the analog phone connected to Zap/1 how do I transfer & hold the caller ? When I hit the flash key, all that happens is the caller hears a beep (sounds like DTMF). But no stutter dial tone on the Zap/1 Port, just continuing conversation with the caller. What could be wrong here? Cheers Jay

...ed that I should change some .h file and recompile zaptel drivers, but could not find the particular piece of code probably because it has changed considerably later on. Also the information I found indicated that the flash time in Europe is generally very short (80 - 120 ms) as compared to the US (750ms ?). I am in the Netherlands. TIA, Hans Feringa

...at the current status of this bug is, but I'm experiencing similar behaviour trying to "emerge sync" with Gentoo's portage. I've seen it on two different machines and only at home .. which may be important because at home I have a commercial Hughes 1.2MB inbound 0.1MB outbound 750ms latency satellite connection. When I bring the laptop to work and connect with a short latency T3 I don't experience the problem. If you need additional info or can suggest useful tests, I'd be happy to play around .. -- Configure bugmail: https://bugzilla.samba.org/userprefs.cgi?tab=ema...

Hi all, For some reason VSAT or Satellite Internet services are not mentioned (or searchable) in this list's archives. I thought I'd let you know that I tested Asterisk using IAX (not IAX2) to make a phone call from an analog phone hooked up to an Asterisk system behind a Linksys router connected to a Gilat VSAT satmodem, and it worked. The "other end" (gateway) is a

Hello. I'm trying to transfer calls from an analog phone (Zap/1, TDM400P card) to some other terminal connected to my Asterisk PBX. If I make a flash hook pressing the phone hangup button quickly it works as expected, I get a new dialtone and the other side is put on hold. But I would like to use my phone's "R" key instead for some different reasons (it's quite easier to use

Hello I need Asterisk (1.4.39.2) to simulate a flash hook (ie. hitting the "R" key on European handsets) so I can put a call on hold, dial a second number, and set up a conference call. By default, linux/include/dahdi/kernel.h sets the flashtime to 750ms, which appears to be too long for European telcos, as they seem to expect a line cut of about 100ms. After editing the DAHDI_DEFAULT_FLASHTIME accordingly, I recompiled/upgraded Dahdi, and ran the script, but Flash() still hangs up the call: =========== [from_fxo] exten => s,1,Wait(2) exten =&...

...write_impl() but this is a slower code path for raw_svector_ostream. Even though the attached raw_svector_ostream patch without virtual write() is still significantly faster then the current raw_svector_ostream since it avoids the double buffering. With this patch, on my system the example runs at 750ms compared with about 1000ms for the current raw_svector_ostream. Would you prefer to continue review on Phabricator? 2015-07-11 20:48 GMT+03:00 Rafael Espíndola <rafael.espindola at gmail.com>: > This makes write virtual. What is the impact of that on the other > streamers? > >...

...ts incremented from efi/main.c by an event initiated from the UEFI API. According to the comment near the #define DEFAULT_TIMER_TICK_DURATION, the jiffies should increment every 50 msec. So a jiffie is not 1 msec but 50 times longer. If this comment is correct, the actual timeout isn't 15ms but 750ms. Or is it 750us because a few zero's are missing? I don't have the UEFI spec so I can't verify. What I did next was just increase the timeout in efi/udp.c:core_udp_recv to 60 jiffies. This made the double-ACK problem disappear completely. >From now on, every packet is received withi...

...efi/main.c by an event > initiated from the UEFI API. According to the comment near the #define > DEFAULT_TIMER_TICK_DURATION, the jiffies should increment every 50 msec. > So a jiffie is not 1 msec but 50 times longer. If this comment is > correct, the actual timeout isn't 15ms but 750ms. Or is it 750us because > a few zero's are missing? I don't have the UEFI spec so I can't verify. > > What I did next was just increase the timeout in efi/udp.c:core_udp_recv > to 60 jiffies. This made the double-ACK problem disappear completely. > From now on, every pack...

Hi I was wondering if there are any special settings that I need to be able to transfer calls. Whenever I press the 'recall' button, I just here a click, and no ring-tone to transfer. in my debug log I get this : -------------------------- Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1 (index 0) Feb 24

We are having line noise issues in our Asterisk to legacy PBX integration. All SIP calls originating from IP phones sound crystal clear. All calls that originate from the legacy PBX (Isoetec 228) and route through the Asterisk and out SIP have a lot of line noise. I believe I have it pinned down to these Blue Alarm errors that I can see on the legacy PBX side. zttool shows no alarm but when I

...rx (for our dyslexic friends) ; signalling=fxo_ls ; ; A variety of timing parameters can be specified as well ; Including: ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce: Debounce timing (default 600ms) ; rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do dist...

...eam. > > > > Even though the attached raw_svector_ostream patch without virtual > write() > > is still significantly faster then the current raw_svector_ostream since > it > > avoids the double buffering. With this patch, on my system the example > runs > > at 750ms compared with about 1000ms for the current raw_svector_ostream. > > > > Would you prefer to continue review on Phabricator? > > > > 2015-07-11 20:48 GMT+03:00 Rafael Espíndola <rafael.espindola at gmail.com > >: > >> > >> This makes write virtual....

...rx (for our dyslexic friends) ; signalling=fxo_ks ; ; A variety of timing parameters can be specified as well ; Including: ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce: Debounce timing (default 600ms) ; rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring...

...0: Sep 20 15:40:31 rmbp kernel: pcm0: Mixer "vol" -> "none": child=0x00000010 Sep 20 15:40:31 rmbp kernel: pcm0: Mixer "pcm": parent="vol" Sep 20 15:40:31 rmbp kernel: pcm0: Soft PCM mixer ENABLED Sep 20 15:40:31 rmbp kernel: pcm0: clone manager: deadline=750ms flags=0x8000001e Sep 20 15:40:31 rmbp kernel: pcm0: sndbuf_setmap 44e000000, 10000; 0xffffff84597ff000 -> 44e000000 Sep 20 15:40:31 rmbp kernel: pcm1: <NVIDIA (0x0042) (HDMI/DP 8ch)> at nid 5 on hdaa0 Sep 20 15:40:31 rmbp kernel: pcm1: +--------------------------------------+ Sep 20 15:40:...

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

...pcm0: Mixer "vol": pcm0: Mixer "pcm": pcm0: Mixer "speaker": pcm0: Mixer "line": pcm0: Mixer "mic": pcm0: Mixer "mix": pcm0: Mixer "rec": pcm0: Mixer "igain": pcm0: Mixer "ogain": pcm0: clone manager: deadline=750ms flags=0x8000001e pcm0: sndbuf_setmap 33c0000, 10000; 0xffffff80e765c000 -> 33c0000 pcm0: sndbuf_setmap 7580000, 10000; 0xffffff80e769c000 -> 7580000 pcm0: sndbuf_setmap 75c0000, 10000; 0xffffff80e76dc000 -> 75c0000 pcm1: <Realtek ALC662 (Rear Digital)> at nid 30 on hdaa0 pcm1: +-----...

Here's a performance testcase for the raw_svector_ostream patch. On my WIndows x64 machine it runs in 1010ms with the current code and in 440ms with the patch applies. Is this OK to commit? 2015-05-02 21:31 GMT+03:00 Yaron Keren <yaron.keren at gmail.com>: > Following a hint from Duncan in http://llvm.org/pr23395, here is a > revised patch. Rather then introduce the