similar to: asterisk as POC(push to talk) server?

Hi I've seen some posts on the list regarding integrating Nokia's PTT (nokia 6020 and nokia 6230i) with asterisk And use * as a PTT server.. So far I was able to have mobile register itself , send an invite to it, and get SIP error 603 (DECLINED) back from it. And ofcourse the PTT sign on the mobile is off. App_rpt , it mentioned that is can do PTT , but it is not clear..

>Hi Jerry > >As others have eluded to, the 'PTT' feature can mean different things to different >people depending on their background.> > >Is it fair to say that you're looking for a one-touch button which initiates a call to >the other end and causes the other end to automatically answer in speakerphone >mode? >If that would foot the bill then have a

Hello all Asterisk Users, This is my first post here. We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server to Asterisk box. Which card drivers do we need? Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Only source proves that this card work with * http://lists.digium.com/pipermail/asterisk-dev/2003-April/000244.html

Hi, i'm trying to get PoC on Nokia Phones to work with asterisk. I think the store-and-forward part could easy be done in the dialplan, but i can't even get the handset to register with asterisk (authentication failed). I'd try'd to find the difference between "pure sip" and PoC-SIP, but didn't suceed. Has someone an idea how to get this to work? Regards, Jay

Hi, I've a samba-3.0.2a running perfectly on redhat-8. Filesystem is EXT3 with ACL support. I can't upgrade the PDC to 3.0.3 or 3.0.4, because shares are only read-only. In 3.0.3 only home share and temp share are writable. All other shares are in read-only mode. Users can see the files/directories, but can't save anything or create new items. When trying to create a new item -

Hello, I am in bit confusion as I am not able to find point of trouble. All extensions are configured same way. All are registered, have same context, voicemail context. There are around 112 extensions. So I am giving example of 2 extensions. one going to voicemail fine and other not. ================================================================================== CLI output: //WORKING --

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

So, incoming calls on zap work just as I expect them - an intro is played, the caller hits 1 for sale 2 for support or dials an extension. I'm using the privacy option for all extensions. When calls come in from zap, they caller is played the priv-recordintro recording, they say their name, and everything happens normally from there on out. However, when the call comes in from sip and

> Date: Mon, 2 Feb 2009 12:56:15 +0100 > From: friedrich.leisch at stat.uni-muenchen.de > Subject: Re: [R] Problems in Recommending R > To: thomas.petzoldt at tu-dresden.de > Cc: "r-help at r-project.org" <r-help at r-project.org>, > useR-2009 at r-project.org, paul at stat.auckland.ac.nz > Message-ID: <18822.57183.637787.426445 at

Hi, We recently created a Ruby on Rails POC, which landed us with a client. More about this initiative at http://www.articlesbase.com/information-technology-articles/cpal-a-ruby-on-rails-poc-1316554.html Thanks, Miti

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

Hi, In case you are interested in testing the Doveadm HTTP API in a bit more simplified fashion I have written a client for it in Python. Feedback is always appreciated (pull requests more appreciated). https://github.com/hnsk/doveadm-http-cli <https://github.com/hnsk/doveadm-http-cli> -Hannu

# HG changeset patch # User Ian Campbell <ian.campbell@citrix.com> # Date 1294742832 0 # Node ID 03997417771be2aa122d7dac31256399f854c5b1 # Parent b279ec057736ead1d7061f8ec8a29a8676dd56b4 PoC: libxc+ocaml: add interfaces to send triggers to HVM guests * ACPI power button event * ACPI sleep button event Required by xapi. Taken from xen-api-libs.hg/xc and adjusted for upstream libxc+ocaml.

Run: # top # vmstat 1 # iostat -x /dev/emcpowerb 1 The latter two you can save to a file. For top, just monitor cpu usage and see if any process is hogging all of it. Colin Laird wrote: > and the fstab settings: > > # This file is edited by fstab-sync - see 'man fstab-sync' for details > /dev/VolGroup00/LogVol01 / ext3 > defaults 1 1 >

Hi All. I am beginning a project of Call center and predictive diales, my call center have 50 operators, I have 50 analog phone line with the company PTT in my country. I have the following questions: 1- Can I to work this project with Asterisk? 2- What caracteristic of hardware need for my servers? 3- For 50 analog phone line what tipe of card digium I need? Thanks in advanced, Regards.

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

----------------------------------------------------------------------- You''ve been invited by Amardeep Singh to use Google Talk. If you already have a Google account, login to Gmail and accept this chat invitation: http://mail.google.com/mail/b-92554dc04c-71510e76dd-tJ2ZYdosQosMitmwHon_o1ooc4E To sign up for a Google account and get started with Google Talk, you can visit:

Ventrilo in version 2.1.4 work properly good under wine. I talk with friend without any problems with sound, but there is one with "Push-to-talk" function (hotkey). When frame with ventrilo have lost focus i can't use this function. I spent a lot of time to googled about it and I found program called xbindkeys. I got idea, that I can use this program to send keystroke to ventrilo.

Hi, is it possible to configure asterisk as a Push-to-task-over-cellular Server? After reading the spec i was expecting that it is just sip, but i can't get my Nokia to authenticate, no matter what settings i use. Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090713/bfb1c06e/attachment.htm