search for: 1500ms

...;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)

...ve done some initial testing, and a SNOM200 SIP phone comes close enough to working that I have not ruled out the idea as completely un-workable. I understand that this is an extreamly hostile environment, the satelite uplink itself introduces too much latency for a standard configuration to work (1500ms) which is most likely where the problems come from. What I'm wondering is if anyone has ever succeded in making a setup like this work. Different protocol (H.323 MCGP etc) or different codecs? Our testing in his driveway revealed the following. 1. Incoming calls achieve a ring almost immedi...

Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different mobile network providers. Nothing. I played with the rx/txgain values from hearing nothing to too loud... I have no more ideas. Marc -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Steve Underwood Gesendet: Donnerstag, 4.

...also read through the lartc documentation/ADSL bandwidth management. I also went to the IRC Channel (no traffic) and tried subscribing to the lartc mailing list (no response). Emailing the wondershaper Script writer. (no response) Ping Times are Not very good. (ping www.yahoo.com) Loaded Average ~1500ms (Using Bittorrent, DL 1 file @~25KB/s) Unloaded Average ~310ms The Link when Unloaded, trying one of those Bandwidth Test (on the internet) unloaded yielded ~230kbit/s. Loaded yielded ~30kbit/s Due to this, Web surfing slows to a crawl, and ftp transfers also slows to a crawl. I''ve als...

What KVMs do you use? I personally like the ConnectPro.

Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. Any

...is to be expected. We are working with satellite links to moving ship platforms (no it isn't military based) and depending on location, sea conditions, and a few dozen other factors, the data link can become somewhat unstable at times. That and the high-latency for IP traffic itself (600ms to 1500ms) over such a link has always presented some challenges for transfering/syncing data. So we accept the fact that even with "acceleration" devices and other methods to lower latency, the bottom line is you can still have some very small to large interruptions in service (and again, depends...

...e/ning/pa/public/system/avatars/34/medium/Screenshot.png [paperclip] -> /home/ning/pa/public/system/avatars/34/original/Screenshot.png [paperclip] -> /home/ning/pa/public/system/avatars/34/thumb/Screenshot.png [4;36;1mSQL (45.3ms) [0m [0;1mCOMMIT [0m Redirected to /users/34 Completed in 1500ms (DB: 47) | 302 Found [http://127.0.0.1/users] Processing UsersController#show (for 127.0.0.1 at 2009-01-08 00:27:19) [GET] Parameters: {"id"=>"34"} [4;35;1mSQL (0.1ms) [0m [0mSET NAMES ''utf8'' [0m [4;36;1mSQL (0.1ms) [0m [0;1mSET SQL_AUTO_IS_N...

...a whole > byte, you can use a single bit. > > that removes the whole NR_CPUS dependent logic. After modifying it to do a deterministic cmpxchg, the test run time of 2 contending tasks jumps up from 600ms (best case) to about 1700ms which was worse than the original qspinlock's 1300-1500ms. It is the opportunistic nature of the xchg() code that can potentially combine multiple steps in the deterministic atomic sequence which can saves time. Without that, I would rather prefer going back to the basic qspinlock queuing sequence for 2 contending tasks. Please take a look at the per...

On Thu, Feb 27, 2014 at 03:42:19PM -0500, Waiman Long wrote: > >>+ old = xchg(&qlock->lock_wait, _QSPINLOCK_WAITING|_QSPINLOCK_LOCKED); > >>+ > >>+ if (old == 0) { > >>+ /* > >>+ * Got the lock, can clear the waiting bit now > >>+ */ > >>+ smp_u8_store_release(&qlock->wait, 0); > > > >So we just did an

On Thu, Feb 27, 2014 at 03:42:19PM -0500, Waiman Long wrote: > >>+ old = xchg(&qlock->lock_wait, _QSPINLOCK_WAITING|_QSPINLOCK_LOCKED); > >>+ > >>+ if (old == 0) { > >>+ /* > >>+ * Got the lock, can clear the waiting bit now > >>+ */ > >>+ smp_u8_store_release(&qlock->wait, 0); > > > >So we just did an

> Perhaps, but then you need to assume that the jitterbuffer can just > throw away the data, and that limits how you can use it. In object- > oriented terms, you might want to pass objects to the JB, and then > call a destructor on them. In C terms, you may want to allocate > frames via malloc(), and then call free() on them later. You might > want to pass in

...I just had this discussion here: http://bugs.digium.com/ view.php?id=6011 Here's the case where you would want it: The theory behind doing that is that in general, you want control to be synchronized with voice. Imagine a situation where you are leaving a voicemail, and you have a big (say 1500ms) jitterbuffer; The VM system lets you press "#" to end the recording. So you say "I'll pay you A million dollars; just kidding!", and press "#"; If the system acted on the DTMF immediately, your message might be seriously misinterpreted. The same concept holds...

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

...o_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 distinctive ring detection on FXO lines ; ;usedist...

...o_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 detection on FXO lines ; ;usedistinctiveringd...

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?