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Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, guys. Have a strange problem with LLVM in my project ( https://github.com/ababo/AntOS). Cannot instantiate JIT execution engine (NULL returns; message: Interpreter has not been linked in.). As you can see from the code I call InitializeNativeTarget. Also I tried to directly include the "llvm/ExecutionEngine/JIT.h" header, but with no success. I link with `llvm-config --ldf...

Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time

...2011/9/2 arrowdodger <6yearold at gmail.com> > >> On Fri, Sep 2, 2011 at 6:15 PM, Semion Prihodko <semion.ababo at gmail.com>wrote: >> >>> Hi, guys. >>> >>> Have a strange problem with LLVM in my project ( >>> https://github.com/ababo/AntOS). Cannot instantiate JIT execution engine >>> (NULL returns; message: Interpreter has not been linked in.). As you can see >>> from the code I call InitializeNativeTarget. Also I tried to directly >>> include the "llvm/ExecutionEngine/JIT.h" header, but with no...

What are the limitations on size of matrix for MDS functions? steve -- Steve Antos Priva-Technologies 847-640-9020 x3114 cell (540)409-1231 [[alternative HTML version deleted]]

Hello everyone, I have come across a bit of an odd problem: I am currently analysing data PCR reaction data of which part is behaving exponential. I would like to fit the exponential part to the following: y0 + alpha * E^t In which Y0 is the groundphase, alpha a fluorescence factor, E the efficiency of the reaction & t is time (in cycles) I can get this to work for most of my reactions,

Hi, I am new to clustering and loadbalancing in apache, What is best way of doing it? How do I do the clustering and what tools do I need to use? Do I have those tools, I use CentOS , Do i have any tools in CenOs which comes default in it? And how do I do apache load balancing? should I rely on apache forums or mailing list or is there any way or tool I can use in CentOS? Can any throw some vague

...old at gmail.com> >>> >>> On Fri, Sep 2, 2011 at 6:15 PM, Semion Prihodko <semion.ababo at gmail.com> >>> wrote: >>>> >>>> Hi, guys. >>>> Have a strange problem with LLVM in my project >>>> (https://github.com/ababo/AntOS). Cannot instantiate JIT execution engine >>>> (NULL returns; message: Interpreter has not been linked in.). As you can see >>>> from the code I call InitializeNativeTarget. Also I tried to directly >>>> include the "llvm/ExecutionEngine/JIT.h" header,...

Hello All, Sorry to resend this email since I really want to know the answer. Did anybody know how I could make Linux Guest OS mount CD automatically with Nautilus? When I exposed a Music CD to my FC5 guest os on Xen 3.0.3, I found the CD wasn''t recognized as a Music CD, it was recognized as a blank CD. In Nautilus, when I clicked the icon of CD, a empty folder was opened. On general

Hello Everybody, Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2: [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729) [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729 [Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing [Dec 28 21:06:00] DEBUG[1756]

Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. -- Tomislav Parcina tparcina#lama.hr

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with

> Date: Wednesday, November 09, 2016 21:25:50 +0100 > From: Antonio Trande <anto.trande at gmail.com> > > On 11/09/2016 09:21 PM, Phil Wyett wrote: >> On Wed, 2016-11-09 at 12:16 -0800, Bart Schaefer wrote: >>> For what it's worth, I just received the auto-notification of >>> having been kicked off the list for excessive bounces. So gmail >>>

Hi, I installed ffmpeg using yum, and then tried to convert a asx video to flv it says unable to find encoder x264, eventhough it is installed and I am able to view it when I check ffmpeg --help.Can anyone tell me how to fix this issue? Thanks and Regards Marky -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everybody, I have set dtmfmode=auto in my sip.conf, but that does not work and I still got the following message: WARNING[4980]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 According to http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode: auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones

Hello everybody, I am not using astdb (no func_db and app_db) so I am wondering why asterisk is always updating it. The interval of the update is not constant. Using lsof, I noted the intervals are somewhere between 1 minute to 12 minutes. The output of lsof says that asterisk, atd and crond processes were just active, just after the hard disk changed the state from standby to active/idle. I

Hi, I have 2 phones AASTRA 57i with Asterisk 1.6. When the internet connection for some reason fall down the 2 phones go to "NO SERVICE", searching on internet i found that this is due to DNS service. Has someone solve this problem? or suggestions? Thanks in advance Antonio Supera i limiti: raddoppia la velocit? da 10 a 20 Mega! Risparmia con Tutto Incluso: telefono + adsl 20

All the 'unknown device' means is that your 'lspci' doesn't know what the card is. That's all. Nothing more. --Rob ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aryanto Rachmad Sent: Tuesday, 4 October 2005 7:43 AM To: asterisk-users@lists.digium.com Subject:

Hello everybody, I have been googling for hours and also searched on http://www.voip-info.org/wiki-Asterisk, but I still can not find any information for the problem I have. So I hope one of you could help me out. I have actually very little experience in Asterisk and also Linux. But by following installation guide, luckily I could get asterisk working. That is only with SIP and IAX channels

flight 21641 linux-3.10 real [real] http://www.chiark.greenend.org.uk/~xensrcts/logs/21641/ Regressions :-( Tests which did not succeed and are blocking, including tests which could not be run: test-amd64-i386-pv 18 leak-check/check fail REGR. vs. 21026 test-amd64-i386-qemut-rhel6hvm-intel 11 leak-check/check fail REGR. vs. 21026 test-amd64-i386-qemuu-rhel6hvm-intel 11