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...ments imply that it should be commented out by default... The main reason I ask is because I usually try to re-compile the kernel to only include the bits that I need, and so I don't include PPP... Regards, Adam Adam Goryachev Website Managers Ph: +61 2 9345 4395 info@websitemanagers.com.au Fax: +61 2 9345 4396 www.websitemanagers.com.au

...ing.... Has anyone got a IP300 phone to display the status ?? Any suggestions for things to look at/etc ?? PS, of course, the current state is that 600 is off-hook and all others are on-hook. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 0000 adam@websitemanagers.com.au Fax: +61 2 8304 0001 www.websitemanagers.com.au

...a new 300G drive, format it from win2k, and then somehow transfer the files from Linux into win2k, without losing any of the permissions / etc which I guess will be somewhat challenging. Any suggestions, or advise would be greatly appreciated. Regards, Adam -- Adam Goryachev Website Managers www.websitemanagers.com.au

Hi everybody! As the easter holidays have now finally arrived and all the exams are over, I will have more time for webgen again! Summary ======= * Changes included in the upcoming 0.4.3 release * webgen repository * Status on 0.5.0 * Detailed information on the new 0.5.0 plugin system Changes for 0.4.3 ----------------- I have collected all mails/forum entries/ML postings in which bugs

you can do that. But are u installing qmail and * on same box. i wont recommend that. i use qmail and *. qmail is strictly for internet email. * is on separate server not exposed to Internet. * box also has sendmail. i hv configured sendmail to use smart host (qmail server). This way its safe and secure. HTH, -B ----- Original Message ----- From: "Ing Isianto Istiadi"

Dear All, I have been running an Asterisk 0.7.1 (patched with various agent applications) server for almost 2 years. We have a data center in the USA and a call center in the UK. All calls are routed to a group of central call queues in the USA. Agents from the data center, call center and from remote locations (London, Scotland, LA, Florida, and Maine) can log in, join the call queue and pick

...ut I need to ensure the module is loaded, the ramdrive exists, and also use dd to copy the image of the drive into the ramdrive (partition, format, and files all in one go), all before the machine bootup commences. Thanks for any suggestions. Regards, Adam -- Adam Goryachev Website Managers www.websitemanagers.com.au

I've just finished a general purpose configuration utility for the GS phones: 1) Generates files from scratch (using MAC), from HTML config listing, or by directly downloading from the phone. 2) Does multiple simulteneous edits. 3) Can reboot as many or as few phones at a time as you like. I would like to offer it to the list, but there are 2 issues: 1) I want to GPL it first, if

I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel

I am trying to download the latest release of Asterisk from: ftp://ftp.digium.com/pub/asterisk/ The link provided by Digium is incorrect for the Asterisk Tarball as there is no such file at ftp://ftp.digium.com/pub/asterisk/asterisk-1.0.7.tar.gz However the links for the Asterisk-Addons and other Tarballs is OK ftp://ftp.digium.com/pub/asterisk/asterisk/asterisk-addons-1.0.7.tar.gz Does anyone

...-24 ii xen-hypervisor-3.2-1-amd64 3.2.1-2 ii xen-linux-system-2.6.26-2-xen-amd64 2.6.26-24 ii xen-utils-3.2-1 3.2.1-2 ii xen-utils-common 3.2.0-2 ii xenstore-utils 3.2.1-2 xen6:/etc/xen# xm info host : xen6.websitemanagers.com.au release : 2.6.26-2-xen-amd64 version : #1 SMP Sun Jun 20 20:51:58 UTC 2010 machine : x86_64 nr_cpus : 6 nr_nodes : 1 cores_per_socket : 6 threads_per_core : 1 cpu_mhz : 3214 hw_caps...

Seems to have struck a small problem.. Using a t100p & Zhone channel bank.... one extension ringing another.....the following will appear WARNING[18448]: File chan_zap.c, Line 2685 (zt_handle_event): Didn't finish Caller-ID spill. Cancelling. if we are using defaultzone=au change it to us and the problem goes away..... any possible solutions ?? Gary .

...thanks for asking :) E100P E400P TE410P I only need one card, in that preference order. Would also prefer if the card is already in Australia. I'm in Sydney/Australia if that matters to you. Regards, Adam -- Adam Goryachev Website Managers Ph: +61 2 9345 4395 adam@websitemanagers.com.au Fax: +61 2 9345 4396 www.websitemanagers.com.au

Hello! I am trying to set up multi-link PPP using two T100P cards in one machine, and 1 T405P card (the 4-port one) in another machine. I have previously been able to get PPP working between the two T100P cards in separate machines.... The 4-port card seems to be my problem currently. I am trying to use the tor2 driver (from a fresh CVS download this morning). When I load the driver (or run

...ust stop asterisk, unload/reload the modules, and restart asterisk. Usually this is enough to fix things... If anyone can suggest some debugging to help find the problem, I'd appreciate it. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 0000 adam@websitemanagers.com.au Fax: +61 2 9345 4396 www.websitemanagers.com.au

Has anyone been able to find a way to disable call-waiting on Polycom phones? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/b7ea7b5d/attachment.htm

Hi, I am mostly happy with my Polycom IP600 but it apparently needs to check the FTP server every minute. I couldn't find any obvious setting related to that behavior in the configuration files. Any idea how to curb the IP600's spurious network activity? Thanks, -- Lord, protect me from your followers.

Hi all, I have call recording enabled via the Monitor command and it seems, the call stops being recorded after the call is transferred. Is this normal behavior? If so how can I continue recording of calls after they have been transferred....

Hi, I intend to let several SIP-phones on my asterisk ring cascaded on incoming calls. First only phone 1 should ring, after 5 seconds phone 2 should ring in addition and after additional 5 Seconds phone 3 should also ring. How can I realize that correctly? Currently I do use Dial(SIP/1,5) Dial(SIP/1&SIP/2,5) Dial(SIP&1&SIP/2&SIP/3) But this seems not to work correctly on

I gave a queue setup like this, but I also have it setup so that if no agents are online, the caller cannot get in but I discovered that if that's the case, the call hangsup on the caller: [soportetecnico] ;Soporte Tecnico exten => 2,1,Playback(${SONIDOS}/transferringcall) exten => 2,2,Queue(Soporte-Tecnico) exten => 2-.,1,Playback(noagents) I want to play a message tothecaller