similar to: failover for voip providers (i.e. Dial() doesn't give enough options)

Hi guys I have tried compiling this attached code, I made all the buffers 320, there is no trace of a 160 buffer, but I get a " SpeexEncoder requires 320 samples to process a Frame, not 160" error. Maybe there's something I'm missing, here's my code: import java.io.IOException; import java.io.FileOutputStream; import java.io.File; import

Hi Everyone, I am trying to pass a variable using the .call files but it turns out blank. Can someone please point out what might be wrong here: */tmp/spool-file.sh* *----------------------------------------------------------------------* echo "Channel: Local/s at callback_leg*1*/n CallerID: \"Call-back\" <123456> MaxRetries: 0 RetryTime: 10 WaitTime: 45 Context:

I'm not mathematically gifted, but shouldn't 957%60 be 15 remainder 57? Google and my desktop calculator certainly think so. So where am I going wrong here? The following code exten => 7,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN}) same => n,Set(myNum=957) same => n,Set(sec=$[REMAINDER(${myNum},60)]) same => n,Set(sec=$[ABS(${sec})]) same =>

Hi, sorry for this noop question, but does anybody know how to cut the first digit of a variable? example: 044612345 should be after cut operation: 44612345 My try in the extension.conf: exten => _0[0-9].,2,Cut(mynum=EXTEN,/ ,1) exten => _0[0-9].,3,Dial(Zap/g1/${mynum},90,T) but it didn't work, my problem is the delemiter, I have no delemiter, the default is "-" but how to

Hello Trevor, I wonder how I can find out for sure what is the H/W version for a PROSODY ACULAB SS7 Card? I dought that I have a ver 1.1 which have may issues with recently made computers. I have a case opened for my problem with aculab but sysdiag shows that I have ver 1.1 and aculab says I have ver1.5! can sysdiag be inaccurate? I tried config summary and I had the same result! wish you

I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single asterisk@home system with a single T1 card. Robbed Bit T1 ami, d4. ------------------inbound call

- Only for sufficiently new xserver's and exa_driver_pixmaps. --- src/nouveau_exa.c | 217 +++++++++++++++++++++++++++++++++++++++++++++++++++-- src/nv_driver.c | 51 +++++++++++-- src/nv_proto.h | 4 + src/nv_type.h | 12 +++- 4 files changed, 267 insertions(+), 17 deletions(-) diff --git a/src/nouveau_exa.c b/src/nouveau_exa.c index 93fc3c5..074a226 100644 ---

I figured it out. The old system (Televantage 3 and 4 I think) has limited specifications on the T1. After setting up the system, I was able to send and recieve calls. I still have some work to do like figuring out faxing and a floating receptionist, but this is a nice start. ----------------------------------------------------Televantage T1 Requirements: Framing: D4 Superframe or Extended

- map should handle this. --- src/nouveau_exa.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/nouveau_exa.c b/src/nouveau_exa.c index b656ca7..20ad380 100644 --- a/src/nouveau_exa.c +++ b/src/nouveau_exa.c @@ -259,7 +259,7 @@ nouveau_exa_mark_sync(ScreenPtr pScreen) static void nouveau_exa_wait_marker(ScreenPtr pScreen, int marker) { -

* Kill the A8+A8 hack. Recent enough X servers (>=1.7) fall back to ARGB glyphs for drivers not supporting A8 render targets. * Kill all the global state. It doesn't matter a lot yet but it might if we get multicard working at some point. * Other random clean-ups with no functional changes. Some numbers from x11perf -aa10text -aa24text -comppixwin10 -comppixwin500: * Before, with A

Hello Mynum: 6001 , Othernum: 6002. I can record as follows. But I do not enter individual records for each internal required. I want to do it more smoothly with a Macro. Thanks. exten => _6001,1,NoOp() exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab) exten => _6001,n,Dial(SIP/6001,20) exten => _6001,n,StopMixMonitor() exten => _6001,n,Hangup() On Sat, Mar 9, 2019 at 6:50 PM

Hi, I am looking at a project which requires VoIP QoS monitoring and failover re-routing to PSTN, without dropping the call ideally. While I have hardware solutions available such as the Quintum Tenor series, I see no reason why Asterisk can't have this feature with some effort obviously. Plus I note it is on the wish list at http://www.voip-info.org/wiki/view/Asterisk+Wishlist (last item

Has anyone tried the new ariavoice $89 VOIP desk phone with Asterisk? ` http://www.voip-info.org/wiki-AriaVoice -- Jim James H. Thompson jht@lj.net

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Polycom 601, hands down. - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Warren Sent: Thursday, June 15, 2006 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best $300 VoIP phone for asterisk? If you had approx $300 per phone as a budget and needed to buy

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

For now pixmaps will only be tiled if driver pixmaps are being used and we're told to with the NOUVEAU_CREATE_PIXMAP_TILED usage hint. Signed-off-by: Francisco Jerez <currojerez at riseup.net> --- src/nouveau_exa.c | 31 ++++++++++++++++++++----------- src/nv50_exa.c | 6 +++--- src/nv50_xv.c | 2 +- src/nv_proto.h | 2 +- src/nv_type.h | 1 + 5 files

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of the

Hi: I use shared internet bandwidth and the calls are very clear from around midnight till about 4 pm when it goes bad after that. Is there a way to boost the internet bandwidth for Asterisk at the peak time? Thanks Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html

Based on the original patch by Christoph Bumiller, but since it depends on kernel support patched I cannot push it yet. The changes are that I enable vblank by default, and offset takes OFFSET_HIGH/LOW instead of something relative to notifier_bo. Signed-off-by: Maarten Lankhorst <maarten.lankhorst at canonical.com> --- diff --git a/man/nouveau.man b/man/nouveau.man index 7c72907..8765569