similar to: A102d sangoma's card and ztdummy

Hi: Please every that work with A102d say how about is it?Is it really difficult to install card for me new in asterisk? Best regards. --------------------------------- Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've found that the FLAC__FrameHeader struct's blocksize member has values limited to the range (0, FLAC__MAX_BLOCK_SIZE] where FLAC__MAX_BLOCK_SIZE is 65535. In the encoder, what determines the size of a block? Cheers, Erik -- ----------------------------------------------------------------- Erik de Castro Lopo -----------------------------------------------------------------

hi has the flac encoder multi-core support (so I mean using 2 or more cpu's simultaneously to encode 1 track at the time, not encoding 2 or more tracks in parallel )? And if yes, how to enable this? thx

On 9/26/07, Josh Coalson <xflac@yahoo.com> wrote: > --- Martin Leese <martin.leese@stanfordalumni.org> wrote: ... > > Where can I find more detail on what is a > > "non-audio" RIFF chunk? > > it is any riff chunk that is not "fmt " or "data" > > > Ambisonic ".amb" files are WAVE-EX files with > > a

Christian Weisgerber <naddy@mips.inka.de> wrote: > #0 0x0000000040d18810 in FLAC__lpc_compute_residual_from_qlp_coefficients_wide > (data=0x49e4c014, data_len=110, qlp_coeff=0x7f7ffffece70, order=1, > lp_quantization=14, residual=0x4fced000) at lpc.c:745 > 745 residual[i] = > data[i] - (FLAC__int32)((qlp_coeff[0] *

Hi: I've configured an asterisk server with A102d sangoma's card and the E1 link.I want to dial outside number using the E1 Link.How can I dial a phone number?Is this true? exten => 123,1,Dial(ZAP/1/phone number) I'd appreciate any help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

--- Harry Sack <tranzedude@gmail.com> wrote: > 2007/7/25, Josh Coalson <xflac@yahoo.com>: > > > > --- Harry Sack <tranzedude@gmail.com> wrote: > > > 2007/7/25, Harry Sack <tranzedude@gmail.com>: > > > > > > > > Hi > > > > > > > > I have downloaded a FLAC file somewhere and when trying to > decode

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

Hi: I want to have conference call service and I have A102d sangoma's card so I install asterisk 1.2.x or 1.4.x? Best regards. --------------------------------- Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I'm interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I

Hi, I don't think Asterisk can do this yet but I was wondering if one could start and stop voice recording on demand (but via voice commands). Our situation: some users would need to initiate a call to a special extension/context dedicated to voice recording. This can be done easily. However, once the call is established, the users can't use their hands but they would need to at least

Howdy, I'm reading your managing software with yum document and wanted to point out that there could be a problem with the examples that show 'su -c" some command"'. Su without the "-" or "- root" (e.g., "su -" or "su - root") will allow current environment settings to be inherited by the new root shell. This could lead to disaster

--- Harry Sack <tranzedude@gmail.com> wrote: > 2007/7/27, Josh Coalson <xflac@yahoo.com>: > > > > > > > But how is it possible then the FLAC encoder allows files which > have > > > a bad > > > resulting MD5 to be encoded? Is it because of the bad ram, ... > this > > > incorrect MD5 is not detected during encoding? > > >

Hi: I want to have conference call(meetme) service with asterisk and 30 users.Now do I use 1E1 or 30 analog lines with due attention to high price of E1 line?And which interface card do I use? Best regards. --------------------------------- Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. -------------- next part -------------- An HTML attachment was

Hi: I installed A102d sangoma's card successfully but Asterisk doesn't answer to incoming call from pstn and console doesn't show any message of incoming call in the other word when I diall the number of E1 I can't connect to asterisk and dial the number of extension. I'd apreciateany idea. --------------------------------- Moody friends. Drama queens. Your life?

Hi all, Are there any issues to be concerned about when calls come in from PSTN to a PRI card and are forwarded back out the same PRI card? Anything different have to be enabled in zaptel.conf or zapata.conf or the Sangoma configs to make this work? What about using .call files that join two ZAP channels? Channel: ZAP/1/4081234567 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Application:

Hey Guy, I have ubuntu 10.04 64bit and compiled dahdi / wanpipe 3.5.19 /asterisk-1.8.x I didn't understand what is the relation between wanpipe and dahdi ? do i need to start wanrouter service ? I am getting weird errors and my system got kernel panic many time when i restart dahdi service. any idea ? what is the startup sequence of all these service ? root at example:/etc/asterisk#

Hi: I want to receive a fax with an E1 link connected to A102d card from a fax machine,but after dialling the phone number, it connects then will be busy.In fact asterisk can't detect the fax.These are zapata.conf, extensions.conf filels and debug in console: extensions.conf: [from-pstn] exten => 9711315,1,Answer() exten => 9711315,2,Wait(10) exten =>

Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console=> notice,error ;messages => notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential

[also posted on Trixbox trunk forum] I am also working with Sangoma directly to debug this, but so far no real luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE 3.2.6 (3.2.7 is out, but nothing has changed that would affect this problem). The system gets about 200 calls inbound on the trunk, which is not very heavily used, and of those calls one or two a day is randomly