similar to: Polycom call drops

Can any explain the following error: Error in if ((seedCount <= seedNumber) && (valueDiff > sup)) { : missing value where TRUE/FALSE needed which I get upon running this script: seedNumber <- 10 seeds <- array(dim = seedNumber) seedCount <- 1 maxValue <- 100 sup <- maxValue / 2 fcsPar <- array(as.integer(rnorm(100, 50, 10))) while (seedCount <=

Can any explain the following error: Error in if ((seedCount <= seedNumber) && (valueDiff > sup)) { : missing value where TRUE/FALSE needed which I get upon running this script: seedNumber <- 10 seeds <- array(dim = seedNumber) seedCount <- 1 maxValue <- 100 sup <- maxValue / 2 fcsPar <- array(as.integer(rnorm(100, 50, 10))) while (seedCount <=

We have tried Asterisk 1.0.9 on FC4 and have never been able to get CAPI (with Fritz card, fcpci) to work properly. Apart from that Asterisk works fine in switching internal calls. But it's useless if we can't make outgoing calls on our ISDN line. We are considering abandoning FC4 for Debian or SuSe. What is the general concensus on the best Linux to run Asterisk with CAPI? /Why Tea

Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on

Chris Mason (Lists) wrote: > Cory Andrews wrote: >> >> >> >> >> IP430, will sit between the IP301 and IP501, IP430 will have dual >> Ethernet, PoE, and full duplex speakerphone. List price (MSRP) $239 >> street price should fall likely between IP301 and IP501. >> > That looks great, the 301 is almost useless due to the lack of speaker

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Jerry Glomph Black > Sent: Monday, January 30, 2006 11:59 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet port > unusable? > > Have just done a

Hi Steve, > .. I'll try to sort all this out, and then I'll attack this > problem. Hopefully, I get it all into svn before the next release of > 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one

Hi; I'm looking for a Polycom distributor in the UK who can supply a small number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ? jd -- John Daragon john@argv.co.ok argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127

This feature also works on the IP301 phones. The obvious caveat is that it is one-way only. Still nice for an "all-page" though. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith > Sent: Thursday, February 09, 2006 12:27 PM > To:

My polycom phone is now hanging at Running "sip.ld". I modified it's config via the web interface to register with my asterisk box. I have tried to restore the default settings wth 468* and it doesn't seem to work. Any ideas? -jonathan

I was wondering under what conditions Asterisk will hand off a call to another switch. I'm trying to verify that my local PSTN's Coppercom switch operates correctly... and wanted to know how to get a call REFER'd to another end-point. Thanks, -Philip

Hi, I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? Be waiting.thanks a lot Marlo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/16cc52cd/attachment.htm

Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as

Can Theora be streamed IP Multicasting? Can Cortado support IP multicast? if not, could some add IP multicasting to Cortado? Find a better answer, faster with the new Yahoo!7 Search. www.yahoo7.com.au/search

at one time, a long while back, kde dt had a 'tea timer' applet for the 'system tray'. do not need to time tea, but do have need for a timer for other uses. anyone know of an app that will run in 'system tray'? tia. -- peace out. tc,hago. CentOS GNU/Linux 6.8 KDE 4.3.4 g . =+= Tired of having your microsoft os hacked? Change to Linux os, used by microsoft

Este correo no es de yanier ----- Original Message ----- From: <samba-request@lists.samba.org> To: <samba@lists.samba.org> Sent: Wednesday, May 11, 2005 1:18 AM Subject: samba Digest, Vol 29, Issue 14 > Send samba mailing list submissions to > samba@lists.samba.org > > To subscribe or unsubscribe via the World Wide Web, visit >

On Thu, February 5, 2015 12:45 pm, m.roth at 5-cent.us wrote: > Valeri Galtsev wrote: >> On Thu, February 5, 2015 10:08 am, Always Learning wrote: > <snip> >>>> I know, I know, everybody is reasonable, it is just I didn't have my >>>> coffee yet... >>> >>> Your logic is amazingly good for a coffee drinker. >>> >> No, I

i want to tag categories to its menuname. i have a csv containing menu item name and in other csv i have a column containing some strings, i want to pick that strings from categories and look into menu items if any menu item containing that string i want to create a new column next to menu item name flagged as 1 otherwise 0 and the only condition is once a menu item flagged as 1 i don't need

Hello, We have a problem on a recent asterisk install with Polycom 30x phones; Sometimes (can not reproduce or find the logic of the problem after one week one analysis), the called party (even incoming or outgoing call) can not hear the calling party, as other flow works (caller hears called). This occurs between 5 and 10% of the time. The configuration is the following: - Asterisk 1.2.9.1 -

hello,everyone! I was setting up ChanSpy in an Asterisk dialplan today and it just wasn't working. Here is the snippet: extensions.conf: [test] exten=>3001, 1, Set(__TRANSFER_CONTEXT=tranfer) exten=>3001, 2, Dial(SIP/3001,10,tr) exten=>2002, 1, Dial(SIP/2002,10,t) exten=>2002, 2, Hangup() exten=>2001, 1, Dial(SIP/2001,10,t)