similar to: Anyone know of a pass through ATA

We are still using 1.4 and were going to start testing with 1.6.0, but then 1.6.1 was released and now 1.6.2 is already in beta 2. That seems like a lot of independent releases to maintain. I read about all the regressions ans hurried dot releases, makes us nervous. How is everyone doing their testing? -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, On our Polycom phones we can not activate the Buddy Watch feature. When you add or edit a contact, the list ends at "Auto Divert".....I know it is the end of the list b/c the down arrow on the right side of the screen disappears when I get to Auto Divert. When I add <bw>1</bw> manually to the speed dial file it doesn't change anything. The buttons work well for

I am trying to figure out which one of our agents is answering the calls. According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the only time the queue_log puts the channel (agent) is during logoff & logon. There is the connect & completeagent message, but it doesn't show which channel (agent) answered the phone. I can't even figure it our cross referencing the

Anyone know of a SIP phone that supports group listen? Group listen allow you use the handset but what the far end says comes out the speaker...it is F802 on a Norstar. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/a219002e/attachment.htm

We have a customer on a wireless connection that has very bad jitter. They can hear people fine, but people have a very hard time hearing them. They are connected via a SPA-2102. It is a SIP client going to a SIP trunk. Something like this in sip.conf [general] would be in effect for all SIP clients: jbenable = yes jbmaxsize = 150 jbresyncthreshold = 1000 jbimpl = fixed jblog = yes I only want

I have an Asterisk box, 1.4.30 with a PRI. A Mitel 3300 is connected to the Asterisk box via SIP trunking. When a user calls from the Mitel through the Asterisk box the user can speak but can not hear the far end. But - when I route the Mitel user to echo() it works, send and receive. The Mitel user also can record and playback greetings. One thing I have noticed is that when the Mitel user

I'm missing something simple I think: I have an spa3102 for which I want asterisk to use the incoming pstn callerid when it sends the call to a local extension (207). callerid works fine for the internal phones (between each other) The spa3102 is picking up the PSTN callerid and displays it in its own status pages Asterisk however, doesnt see the callerid at all. The spa3102 is set to:

Hello, I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that the MWI indicators will never go off (The blinking red light and envelope in the LCD). I have tried to upgrade to 1.4.14 and all different SIP versions on the Polycoms. I am now at 1.6.7 Here is the SIP Message that turns on the lights: Scheduling destruction of SIP dialog '

Does anyone know if you can have multiple TE110P cards in one chassis? -Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061011/adbceeb5/attachment.htm

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

UPDATE - This is really really bad - check out the paypal phishing example on my blog already using Cyrillic characters http://blog.collins.net.pr/2009/12/de-latinisation-of-web.html Please forward to everyone in a position to stop ICANN, i cant believe they didn't think of this in advance. Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at

Hello, I have been tearing my hair out on this issue for 2 days, any help would be appreciated. We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch There are two VLANs, 1(data) & 50(VoIP). When Polycoms are connected to the switch with VLAN 50 hard coded in the config they grab a DHCP address from VLAN 1, the PVID for the switch port. The ports have membership in

Hello list, anyone here using Asterisk together with HTB for queing incoming and outgoing packets ? I've tried to subscribe myself to the Mailinglist of the Linux Advanced Routing & Traffic Control project, but I get no confirmation. This list seems dead. It seems my test case with HTB is not giving any noticeable results. Can I ask questions on this mailinglist ? Perhaps you can

Hi. Thanks for answering. Le 2015-09-23 17:34, Dominique Ramaekers a écrit : > Linux has two methods to use ntp: > > ntpdate: > It will run once at boot time to sync time. (This is probably > installed on your system) > It will not run after suspend and resume... => no correction Nope. This is not installed on my system. > ntpd: > Continuously adjusts time. The

Hi, has anyone successfully used the acts_as_threaded plugin with postgresql? I''m using rails 1.0 and ruby 1.8.4 on linux. Following the screencast on http://www.railtie.net/articles/2006/02/05/rails-acts_as_threaded-plugin , I got to where we''re ready to create our first post, having made the changes to controllers/posts_controller.rb, views/posts/_form.rhtml,

Hi, On windows, get_peername works both with connections obtained via start_server, and connections obtained via connect. On linux, get_peername seems to only work with connections obtained via start_server. (I''m using svn HEAD.) I''m wondering which is the intended behavior? (I realize get_peername is less useful for connections obtained via connect, since one should already

Le Thu, 24 Sep 2015 05:32:45 +0800, Bill Kenworthy <billk@iinet.net.au> a écrit : > Look into the "panic" option to ntpd - once the gap gets to big (such > as when the VM is suspended for a few hours) it goes into freewheel > and doesn't sync - its in the ntp docs. My use case is when rebooting the host (after a kernel update, for instance). The gap is about 2

I would like to be able to place a call on hold via the manager interface and be able to retrieve it. The user can click a button in the Order entry form to put the caller on hold when they are looking up information. It saves them from having their hands leave the keyboard and press hold on the phone. I don't see 'hold' & 'retrieve' commands for the manager interface.

I have a virtual server with godaddy but can not compile DAHDI as it complains that I do not have the correct kernel source. The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686: Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest version Nothing to do uname -a returns: Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07

I have written a function that goes through a database and calculates various metric scores and allocates them to a data set. For 400 of the 500 sites that I calculate these metrics for works fine and allocates the scores into the appropriate column. For some reason, some sites I run into the below problem: Here is what I am doing: > names(orig.metric) [1] "BenInsect"