search for: paulh

Sorry for the repetition, unless I've got bad deja vu this questions been asked before but I couldn't turn up an answer on CRAN. Is there already any code in existence for local dependence measures such as Moran's I or Getis and Ord G? Also, S-Plus has a number of interstingly named Cluster Algorithms based on some previous stand-alone fortran algorithms (agnes, daisy etc.) which

...s to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH

...e IM features just kinda worked, without any extra configuration. Messenging and presense. I've heard of folks trying to interface this functionality with MSN Messenger and such, without much success. Can't help you much there. - Jeff On Fri, 17 Sep 2004 11:12:03 +1000, Paul Hales <paulh@adairs.com.au> wrote: > The Polycom phones have an instant messaging function - any idea what > is required to make it work? > > PaulH > Adairs _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mai...

...ir dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code was unproductive... Any ideas? PaulH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060201/8db8e6e8/attachment.htm

Hi, If I setup an IP PAX gateway to handle VoIP calls to a traditional phone line, I am wondering how each VoIP call to the PSTN connection get charged by a local Telecom. Thanks Sam

Hi, I am trying to implement the callback feature of our old phone system. This feature may go by a different name in asterisk? It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B

...ff -u -r1.1 util.c --- util.c 2002/03/19 15:01:28 1.1 +++ util.c 2002/03/19 15:29:32 @@ -483,7 +483,7 @@ { pid_t newpid = fork(); - if (newpid) { + if (newpid > 0) { all_pids[num_pids++] = newpid; } return newpid; -- Paul Haas, paulh@hamjudo.com http://hamjudo.com

Is there a way to specify multiple email addresses in voicemail.conf for a specific user? I seem to remember that it was possible, but can't remember the character to separate the email addresses. (I tried '&', but that didn't work...) later, PaulH

...rial Circle, Suite D Longmont, CO 80501 Available via SameTime Connect within Philips, n9hmg on AIM perl -e 'print pack(nnnnnnnnnnnn, 19061,29556,8289,28271,29800,25970,8304,25970,27680,26721,25451,25970), ".\n" ' "There are some who call me.... Tim?" Paul Haas <paulh@hamjudo.com> Sent by: rsync-admin@lists.samba.org 03/21/2002 08:44 AM Please respond to paulh To: rsync@lists.samba.org Jie Gao <J.Gao@isu.usyd.edu.au> cc: (bcc: Tim Conway/LMT/SC/PHILIPS) Subject: Re: rsync & io Classification: On...

Does anyone know of any startups using Asterisk? What about established companies? Ones that are hiring would be nice.... :) Doug.

Any idea how to read an external file, grab some stuff and push it back into an Asterisk variable? I can do it the other way with: system(echo "${UNIQUEID} =>" >> /home/ast/curr_calls) but I'm a bit stumped on how to go the other way around.... much thanks, Paul Hales

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

In "The future of Telephony", it says "... We should also note for security's sake you should always make sure that your [incoming] context never allows outbound dialing. (If by chance it did, people could dial into your system and make outbound toll calls that would be charged to you!) The book was demonstrating using a PSTN environment and the zapata.conf was something like:

Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal ____________________________________________________________________________________ Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545469

Dear all, I've recently installed Asterisk and am trying to understand where exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy? (or only as a register server?) I am specifically interested in SIP. Could anyone perhaps point me out to a diagram with SIP users and Asterisk to better understand how I should set up my network? Thank you

Hi List, Does anyone have experiences to relate on the various Asterisk-based PBX appliances out there? Like the Aastra 160, Digium S844i, etc. Do the Epygi Quadro and Grandstream GXE also use Asterisk? Thanks, Femi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy

Hi, all I am doing "prrof of concept" system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be

I am trying to get the presence/hints/BLF working along with Realtime SIP but I never get any "busy" notification. core show hints always shows the realtime sip user as idle. I have tried setting call-limit to various values, including 1 but nothing seems to help. I have tried limitonpeers both yes and no. Anybody got any other ideas? I do know the hinting is working as I can

...des' over an isdn2 link - such as *#24# - to activate call forwarding. But it doesn't work. I have tried sending it as a straight dial, and also as a DTMF string...but no luck... I spoke to a telco tech and he said I had to send a facility code....huh? Anyone with any ideas on this one? PaulH