similar to: enhanced speed dial

Hi, new to this list, so a little prelude to my issue with Dovecot. We have been using JavaMail against Cyrus for ages, and developed Webtop, a huge Java web collaboration application running on them in production in various installations for all this time. Recently we had to run the same software against Dovecot pre-existing accounts running on Nethesis NethServer solution. After some time of

Noob Question 31,265,232 if I''m searching on an object, say order, that has many "order_lines" and I want to display order lines by Quantity ( an attribute of the order_lines ) descending how could I do that without having to do a find() with :order, but something like; Order.order_lines.each do |ol| <!-- code to diplay the line --> how can I determine the field

Hello, I can see that there is an implementation/emulation of ddi_pathname in DTrace, but I''m a bit confused about the capabilities and invocation of this function. I would like to diplay the path to the block device from bdev_strategy and other io:genunix::start probes. If someone is familiar with ddi_pathname, could you please provide an example invocation? Thanks, Michael This

Yes, I have both Call Manager and Call Manager Express integrated with *. Prior to Call Manager 4.0 you would need to perform an H.323 integration with *. As of CM 4.0 Cisco supports SIP trunking so this would be the preferred method of integration. This config is on http://www.voip-info.org Seems like the site is having problems now otherwise I would have provided the direct link. I also

Hi, Recently I been using EyeBeam with Asterisk. I can make calls with video, I'm using two PCs with EyeBeam but I noted that I can't enable only one client to stream video, I mean if I start streaming video on one client the other one doesn't receive any video until it starts streaming video itself. So I cannot have unidirectional video calls. I remember this problem with

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

hi, I''m new on this list and i need help from experienced users to give andswer for some questions. I sen up a dom0 (debian) and would like it to be very minimalist, just xen tools, some gnu utils and other usefull utilities. This dom0 will be used to virtualize many hvm OS like windows and some other *nix paravirtualized. Till there are no problem to view/connect to console of paravirt

Hi, I have configured a certified asterisk 13 server with chan_mobile and res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone client. Now the problem is, using the hardphone I'm able to call the softphone and hear everything properly. But when I call from the hardphone to some number that has to be dialed via chan_mobile, I'm not able to hear what the other side says (I

http://sip.house.com.ar/operator Its a server/client combo that displays the status of your Asterisk PBX in a web browser in real time. You can also perform some actions. Hang-up channels and Transfers via drag and drop. The difference with other similar tools is that it displays status in real time (no refreshing necessary), and its graphically appealing. It's a work in progress... so

I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented

Just upgraded to cvs Head this morning and noticed our voicemail notification (via email) is failing with: Jul 1 07:48:38 WARNING[1217669936]: app_voicemail.c:837 sendmail: E-mail addres s missing for mailbox [3000]. E-mail will not be sent. However, a valid address in voicemail.conf has been working just fine until now. Sendmail is running, etc. If I add a "second" email address

Hi, I'm banging my head over this. Usually, I'm using a SIP hardphone feature called "Call Pickup Starcode" to enhance BLF with Directed Call Pickup : basically, SIP hardphone (here a Thomson ST2030S) is configured to send an INVITE message whenever a BLF is pressed while blinking. The INVITE is build with the extension number (attached to the BLF that was blinking and pressed)

Hi, I've been asked sometimes to tailor call history features embeded in SIP hardphones. For example, a cutomer wanted internal call to be taken out. Another wanted calls to sorted according specific criteria. 1. Have you identified a phone offering the possibility to display as Call History, an XML list produced on a distant web server ? With this feature, you would simply have to tell the

Besides the codecs that * supports. Is there any ISAC implementation for asterisk available? This is to be used mainly with softphones, i haven't seen any hardphones that support this codec. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi, The case I'm working on is : - a call comes from PSTN to a given extension (say 122) - a user picks the call up (dialing *8122) from another extension (say 240) using a SIP hardphone - the hardphone (he one with 240 extension) displays the dialed string (here *8122) instead of original caller-id. This is logical but I would like to change this default behaviour so that original

Hi, I'm looking for a few pointers on using ATA to connect Asterisk to the PSTN. Basically, I'm running a Hosted PBX service, and in urban centers I can usually get SIP or PRIs. Since I sell my customers SIP hardphones, the data flow is like this: Customer's SIP Hardphone ---- My own Asterisk ----- Outside lines But when it comes to smaller villages (I deal with people in tiny

Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34

I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give.

Hello, How would you implement free sitting ? The idea is to offer teachers the ability to share the same desk and hardphone : for instance, Mr Foo is teaching mechanics on mondays while Mr Bar is teaching english on wednesdays. Each has his own extension but use the same hardphone. 1. Does a program check a calendar or database somewhere to allocate a phone to a user (as teachers schedules are

Hello, we want to setup the following scenario: - each user has a softphone AND a hardphone - the softphone is started with the operating system - the hardphone is connected all the time using SIP - only ONE extension for each user Both phones should ring when the user is called. We've setup an asterisk 1.4.18 and at the moment only the last registered client rings. In Asterisk 1.2 the