similar to: Re: Asterisk-Users Digest, Vol 20, Issue 91

Hi John, Kevin, Tor and Wiley (and everyone else) - >> I guess the phone just doesn't register as busy when there is only one >> call on a line. It has to have two calls on a line appearance to >> register as busy. Has anyone figured out how to disable this hold >> feature and just have the second call go to the second line, the third >> call to the third line,

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not

Hi - I've just been testing out the latest CVS HEAD (as of about 10:00a EDT today). I'm getting some weird errors. Calls from one sip phone to another have OK audio in one direction and highly scrambled audio in the other direction. The console shows this error repeated ad nauseum during each call: Jul 18 16:08:03 ERROR[22941]: utils.c:509 tvfix: warning negative timestamp

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

Hi - > I am not sure what I did but blind transfers do not work. The Polycom does > not allow me to dial the extension of the person I want to transfer to after > I hit: > > transfer -> blind I would strongly suggest getting the latest firmware, and using the sample configuration files with that firmware to set up your phone. This SHOULD work. If it still does not work

Hi - We have a couple of large spaces that we'd like to cover with dedicated conference units like the Polycom Soundstation IP3000. We're concerned about adequately covering the spaces, though, one of which is very long and narrow. I wanted to get external add-on microphones for the IP3000, but I've found that unlike some of their other conference products, it does not have this

Hi - I've told lots of people about the Flash Operator Panel before, but I've never actually used it myself. I've got it all set up nicely, but I can't seem to authenticate to the asterisk manager (which is running on the same box). When I set the op_server.pl to give debug messages, it shows that it can reach the asterisk manager, but cannot authenticate: ** Asterisk

Hi - Thanks for the fast response! > Try changing your dial statment to use a group instead such as > Dial(Zap/g2/3055551212) Same result: Spawn extension (from-sip, 000, 1) exited non-zero on 'SIP/2000-8dca' -- Executing Dial("SIP/2000-a4f1", "Zap/g1/19142246402") in new stack Dec 14 22:11:09 NOTICE[1506]: app_dial.c:800 dial_exec: Unable to create

Hi samba List, I am finding a strange problem between a mount samba directory. Any clues why this is happening? I have two servers. The samba server is a Western Digital World Edition II (2) server. The samba client (server) is running Ubuntu-kernel linux-2.6.24-22-generic The samba server is exporting the directory: /shares/internal/Music/ The client is mounting the exported directory to

> > >Noah, I could really use 1.3.4... however, a better question might be >how are *you* getting 1.3.4? I can't seem to get this from anyone, >including my reseller. > > I know how you feel! Polycom has made a product that sells like hotcakes, but they don't seem to want to support it at all, even when you do try to go through the proper reseller channels. I

Hi Dan - > Got this warning when upgrading to 1.2.3 even when using the most > current asterisk-addons and even svn asterisk-addons. > > WARNING WARNING WARNING > > Your Asterisk modules directory, located at > /usr/lib/asterisk/modules > contains modules that were not installed by this > version of Asterisk. Please ensure that these > modules are compatible

Hi Samba List, I am finding a strange problem between a mount samba directory. Any clues why this is happening? The server side is WD MyBook World Edition II and the export directory is: /shares/internal/Music/ on the client site I am mounting the directory to /mnt/mybook-music the client is an ubuntu server $ uname -a Linux tsunami 2.6.24-22-generic #1 SMP Mon Nov 24 18:32:42 UTC 2008

Hi, Our discussions about 64 bit R has led me to another thought. I have a nice dual core 3.0 chip inside my Linux Box (Running Fedora 11.) Is there a version of R that would take advantage of BOTH cores?? (Watching my system performance meter now is interesting, Running R will hold a single core at 100% perfectly, but the other core sites idle.) Thanks! -- Noah

Hi - I found the "maxusers" defined in meetme.c, but I'm not sure how this value is set. Does anybody know if one can limit the number of users permitted in a meetme conference? I know there's MeetmeCount(), but I'd rather avoid the dialplan logic and just set maxusers instead. Thanks, Noah

Dean - > if you are having these troubles and everyone keeps advising you to use > asterisk@home why aren't you doing this? Not to be rude, but I believe you are the only one who suggested he use asterisk@home. asterisk@home is a great (actually stupendous) product for those that want to have a PBX with a GUI up and running within a few minutes. It does not, however, force users

Hi - I just deployed a system using IMAP Voicemail. During my testing, voicemail worked fine. I could check vm from the phone, and the messages would get marked as read, or I could read the messages in a mail client, and the phone's mwi light would turn off. Very neat. I'm not exactly sure when things got munged up, but something broke. I can record messages with Voicemail(), but now

Because you learn how asterisk works by looking at the AMP dial plan fast. With phpconfig you have the ability to view each of the aspects and learn a lot faster than reading the wiki. It also gives you an understanding on how a professional dial plan should be laid out. If he doesn't even understand how to do grouping he obviously hasn't even read the most basic sections of the wiki.

We have two small business that run out of our office. One business has 3 phone lines, and the other has only one. In a perfect world, Asterisk would indicate WHICH line (or group) the outsider caller called, so that we would know which way to answer the phone. The incoming calls would go through a voice mail menu first - different for each company. (Press one for sales, two for

Hi there, might somebody please recommend the best GUI rsync client for XP please? Cheers, Noah