similar to: Zaptel timing on TE405P

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

The raid card is an Adaptec 2100S. -----Original Message----- From: Josh Paetzel [mailto:josh at tcbug.org] Sent: Saturday, November 11, 2006 9:35 AM To: tclug-list at mn-linux.org; pjcrump at bitstream.net Subject: Re: [tclug-list] Drives Not Recognized on Dell Poweredge 1550 CentOs install On Saturday 11 November 2006 01:20, Phillip Crump wrote: > I have a new (used)Dell Poweredge 1550

On Thu, Nov 6, 2008 at 7:50 PM, Anthony Francis <anthonyf at rockynet.com> wrote: > http://en.wikipedia.org/wiki/Jacque_Fresco > > A resource based economy. > > Greg Woods wrote: >> On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: >> >>> Gotta love this list being farmed for spammers now. I am sure they call >>> it targeted delivery or

patents, but it's full of legal terms. Maybe anyone can comment? http://www.europarl.europa.eu/commonpositions/2005/pdf/c6-0058-05_en.pdf Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Hi, I meant - if you have more specific questions - please ask them. And writing back to ML would be desirable, because this info might be useful for other people. I can't give you my dialplan, because it's too large and probably useless without lot of external configs. I can just tell you where to look in info, and if you don't have something working as expected - you're welcome

Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

Hi, Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

On 12/17/07, Jared Smith <jsmith at digium.com> wrote: > On Mon, 2007-12-17 at 12:00 -0800, shadowym wrote: > > I do wish Digium or whoever tests this stuff had a more reliable way of > > testing software releases rather than relying on feedback from the > > community. Fonality, for example use what they call a "hammer" which sounds > > to me like a

Hi, I just had my production box deadlocked - no calls could go trough, CLI didn't load. Last lines in log were: [Jan 11 09:15:43] VERBOSE[7265] logger.c: -- Executing [28901 at local_dial:40] GotoIf("SIP/204.11.200.152-c0070ed0", "1?41:57") in new stack [Jan 11 09:15:43] VERBOSE[7265] logger.c: -- Goto (local_dial,28901,41) [Jan 11 09:15:43] VERBOSE[7265]

Hello, I'm trying to find simple MP3 player in flash, to integrate it with call recordings. My requirements would be: * simple UI * buffering (would be nice) * slider * volume control * support of 8kHz stereo mp3 * javascript access to seek/position * free for any use (GPL, MPL, MIT, BSD) So far I've found that JWplayer[1] does great with my recordings. However it's not small in

Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error" I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be

Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in

# moving to -users as this belongs there. It is a nice idea to run several Asterisk processes simultenously, it will defineately help with multithreading. However I would suggest trying less instances - that would perhaps give greater benefit, as Asterisk has it's own threading. For example 8 instances of Asterisk / 4 instances.. However, in this case - if You go for splitting everything up,

Thanks a lot for the help. But if Asterisk has private IP address and the only way to access it from remote sites is to have vpn connection to the site that asterisk existed (the site has vpn), then how that will happen from the Mobile to be able to run the softphone from the mobile? Any help? Regards Bilal ----------------- I installed out of curiosity today, and guess what? You can do SIP over

Does anyone know where i can find out how powerful a machine has to be to handle a certain amount of call volume? Eg, 2Ghz is enough processing power to maintain 100 calls at a time. 4Ghz is engouh to process 250 calls etc etc. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Has anybody seen that Audiocodes gateway is replying with "486 Busy here" when it's actually not (last call ended ~15 seconds ago). I see this quite often. From other logs i see that previous call ends at 11:13:01, then app_queue tries to dial at 11:13:14 and fails numerous times, before succeeding at 11:14:02 I have attached sample SIP debug log: Any ideas what i could

Hi all, I decided the best way to get to know * well is to do it from scratch. Having read the majority of the "Asterisk: The Future of Telephony" I am now attempting to compile zaptel 1.4.0 and am receving the very same series of errors mentioned in this post on the forums: http://forums.digium.com/viewtopic.php?t=13619&highlight=zaptel1+++zttranscode++error However, there has

I'm getting the following error trying to compile zaptel 1.4.0-beta2 on my RH9 (2.4.20-8smp) system. Can anyone shed some light on this? zttranscode.c: In function `zt_tc_mmap': zttranscode.c:387: warning: passing arg 1 of `remap_page_range_Rsmp_d8cd9cb2' makes pointer from integer without a cast zttranscode.c:387: incompatible type for argument 4 of

IQ Labs announces the release of PBX Testing Framework. This software is intended to test existing call-center PBX, and is distributed under GPL license. Currently it allows SIP testing, but implementing IAX (and even Zap) shouldn't be a problem, as the framework is based on Asterisk, and can do anything the Asterisk does. Please see README file included for configuration and scripting

static void senddialevent(struct ast_channel *src, struct ast_channel *dst) { manager_event(EVENT_FLAG_CALL, "Dial", "Source: %s\r\n" "Destination: %s\r\n" "CallerID: %s\r\n" "CallerIDName: %s\r\n" "SrcUniqueID: %s\r\n" "DestUniqueID: %s\r\n" "CDRUserfield: %s\r\n", src->name,