search for: mattrubi

Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over

How do I set up an existing call to dial out to a new terminal which is included in a conference with the two existing legs of the call? When the dialplan executes the Dial(<terminal>) command, control does not return to the dialplan until the terminal disconnects, after which it's obviously too late to conference it. Is there a conference command or option that lets the dialplan dial

I currently have an AGI that calls the Festival text2wave app to write a wav file that my dialplan plays into a call with the Background() command. But the voice sounds terrible: like SAM, the 1980s 6502 voice synthesizer. I tried to slow it down by calling (text2wav -eval "(Parameter.set 'Duration_Stretch 1.4)" -scale 2.0 [...]), but it still sounds like it's talking while

Is there any calendar client that can point at OX for calendar data, which client can display multiple calendars simultaneously as *overlapping layers* in the GUI? With UI to de/select calendars from view, one by one. That is, a single grid of days displayed, with the events in each day displayed in the same day's view list, as if the layers were all events in a single calendar. And is there

OpenWengo has just released WengoPhone v2.1.0: http://www.openwengo.org/index.php/openwengo/public/homePage/news?payload[newsId]=0 . Has anyone had success (or notable failures) using it as a client for Asterisk? Any advice on integrating it into dialplan, apps, config DBs, etc? -- (C) Matthew Rubenstein

created you must place it in your web directory on the server. =20 I chained the command and also wrote the output to an xml file in the web directory. The command looks like this: =20 'php /etc/asterisk/directory.php.txt > /var/www/html/directory.xml' =20 System Speeddials using Services Button =20 =20 For speed dials I modified the php code to look to a specific file in the

Do you know where to find clear developers' guides (with some examples) for developing apps that run *on* Cisco 79xx phones (especially the 7970)? Examples that can run against Asterisk (not CallManager) with SIP firmware (not SCCP), and/or LDAP directories (or other open servers) would be best. On Sat, 2007-09-01 at 12:00 -0500, asterisk-users-request at lists.digium.com wrote: > Date:

Hi List, Does anyone know of a good dual wan router that can handle SIP well and can failover between connections if there is a SIP issue on one of the lines (meaning there still is a connection however there isnt enough bandwith or sip packets arent going thru etc.) ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

If you write a HowTo, would you please insert it into the wiki at http://www.voip-info.org/wiki/index.php?page=LDAP ? Thanks. On Sun, 2008-03-16 at 07:09 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Sat, 15 Mar 2008 18:20:32 -0200 > From: "Gonzalo Servat" <gservat at gmail.com> > Subject: Re: [asterisk-users] LDAP > To: "Asterisk Users Mailing

Hi all, I have successfully compiled and installed Asterisk on an Alix board (AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian variant). I'm using it at home for a month. I wondered how much it could be loaded, so I tested it with pbx-test: I could place up to 15 simultaneous SIP calls before it got no more responsive. All in all a good, stable and cheap solution for home

I just read on Slashdot (at http://yro.slashdot.org/article.pl?sid=07/12/16/222243 ) that Trixbox "has been phoning home with statistics about their installations", as a Trixbox user exposed in "Trixbox Phones Home" at http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home . -- (C) Matthew Rubenstein

file is created you must place it in your web directory on the server.<br> &nbsp;<br> I chained the command and also wrote the output to an xml file in the = web directory.&nbsp; The command looks like this:<br> &nbsp;<br> &#8216;php /etc/asterisk/directory.php.txt &gt; /var/www/html/directory.xml&#8217;<br> &nbsp;<br>

The US has a Do Not Call list to which people can subscribe to prevent being called by advertisers. Federal laws (strengthened by some state and more local laws) assign penalties for calling people/phones on the DNCL. Is there a query gateway that Asterisk (or an app using Asterisk) can filter through to ensure a number is OK to call (not on the list) before calling it? -- (C) Matthew Rubenstein

I've got a Cisco 7970 that's not completing its network registration to Asterisk. The "Registering" message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes "Updating CTL" then "Loading...", then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed

On Sat, 2007-07-07 at 08:39 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Fri, 06 Jul 2007 12:02:53 -0600 > From: Stephen Bosch <posting at vodacomm.ca> > Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID:

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

I have an app that generates callfiles in the outgoing queue, which connect a channel to an AGI (Perl script) at an extension. The AGI calls the Dial command over a SIP channel. Sometimes I need to stop the outgoing calls after the requests have been made. I delete the callfiles from the outgoing directory, but there are still some calls "in the pipeline". Especially if Dials failed at

I've got an Asterisk (v1.2.11) installation running, but it doesn't seem to have the Meetme() app. At the CLI, I type Meetme , and it responds No such command 'Meetme'; meetme doesn't show up in CLI show modules . I'm running a SIP-only server at a datacenter where I can't add Digium (or any other) HW, and am running under CentOS. There is an /etc/asterisk/meetme.conf

I'm trying to use Asterisk (v1.2.11) make a callback that dials both legs of the call into a Meetme() room together, but I keep getting "conf-invalid" messages. I created a callfile (/var/spool/asterisk/outgoing/out.call) that specifies a Local channel (extension) which contains a Dial() command to the "dialer", and an extension which contains a Dial() command to the

I have a SIP carrier which accepts only G729 connections from my Asterisk server. If all the server does is Dial() (out) two legs of a call which are natively bridged, with no processing the media (and no DTMF detection, etc), do I need to install a G729 codec of my own? All the media from each leg connected to the other is already encoded into G729 by the SIP carrier from which it's coming