similar to: Tests in VMWare (was: Re: asterisk-users Digest, Vol 44, Issue 104)

I'm just wondering if any one else has tried to successfully install Asterisk on Ubuntu inside VM. I've installed Ubuntu without incident or error. Even the install of Asterisk is relatively straightforward as it is maintained in one of the repositories. But when I attempt to start Asterisk I get a nice Segmentation Fault. I've narrowed down the problem somewhat. If I disable modules

Forgive me if this has been covered before. I did search but I was unable to find a reference. I am curious to know more about the possibility of using SQL to store voicemail as well as having more than one voicemail system accessing a central SQL database. Any information would be appreciated. Thank you all, in advance. -- Ein Bielaczyc <ebielaczyc at gmail.com> NOTICE: This E-mail

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

such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with

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

static void senddialevent(struct ast_channel *src, struct ast_channel *dst) { manager_event(EVENT_FLAG_CALL, "Dial", "Source: %s\r\n" "Destination: %s\r\n" "CallerID: %s\r\n" "CallerIDName: %s\r\n" "SrcUniqueID: %s\r\n" "DestUniqueID: %s\r\n" "CDRUserfield: %s\r\n", src->name,

About the IVR, are u using Asterisk? Regards Bilal > ------------------------------ > > Message: 17 > Date: Wed, 18 Feb 2009 12:23:41 +0200 > From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> > Subject: Re: [asterisk-users] Credit Card processing > machines > To: asterisk-users at lists.digium.com > Message-ID: <20090218102341.GD21440 at xorcom.com> >

Same question for asterisk-users as well: ----- Forwarded message from Tzafrir Cohen <tzafrir.cohen at xorcom.com> ----- Date: Wed, 7 Mar 2012 21:14:04 +0200 From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> To: asterisk-dev at lists.digium.com Short version: it's now time to remove. Anybody actually uses latest DAHDI with RHEL4? See inline, On Thu, Dec 29, 2011 at 07:42:39PM

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

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP

---------- Forwarded message ---------- From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> Date: Oct 4, 2007 12:56 PM Subject: Re: [asterisk-dev] chan_h323 and chan_oh323 compatibilities To: asterisk-dev at lists.digium.com Hi On Thu, Oct 04, 2007 at 11:46:30AM -0300, Caciano Machado wrote: > I'm receiving a lot of warning messages from my Asterisk > 1.2.5/chan_oh323 every time

Hi folks Xorcom Rapid 1.0 is avilable for download Q: 1.0? A: Sure, better than 0.9.1: * Asterisk 1.0.5 * Base packages upgraded * Built with SpanDSP support * Improved Zaptel detection * ast-cmd with some useful command-line abilities provided * ssh installed by default * putty.exe is included on the CD * music-on-hold files removed due to potential licensing issues

Thank you Tzafrir. I had been using different users in earlier attempts to make this work. Decided to try everything where root is the only user, simply to verify it's working. For problem 2, where asterisk is writing to the log but doesn't seem to receive the SIP packets even though tcpdump indicates they are making it to the box on 5060, I am starting asterisk while logged in as root.

, address 2 Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.344611] ERR-xpp_usb: XBUS-00: x= usb_listen: usb_submit_urb failed: -19 Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.353325] INFO-xpp: XBUS-00: [] D= isconnecting Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.353373] Zaptel: Master changed = to XBUS-00/XPD-01 Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.353382] Zaptel: Master changed = to

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

Okay, i'll move my discuss to asterisk-users. Thank you. On 1/17/07, Tzafrir Cohen <tzafrir.cohen@xorcom.com> wrote: > > > On Wed, Jan 17, 2007 at 04:39:03PM +0800, ??? wrote: > > Jonson Player wrote: > > > Hello, I intend to buy a FXO/FXS device from Linksys. > > > I'm thinking about SPA3102. What you guys thik about it. > > > Is ok, is

After a reboot, I can't reproduce the problem anymore which is quite frustating. 2011/6/14 Tzafrir Cohen <tzafrir.cohen at xorcom.com> > On Tue, Jun 14, 2011 at 03:44:32PM +0200, Olivier wrote: > > Hi, > > > > I'm using a two-years old installation script for the first time on a > > Squeeze (linux 2.6.32) platform. > > For an unknown reason (might

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

one under my default context at extention.conf. And what is [pbx_config]? Thanks Eyal -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, June 25, 2010 4:05 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Is there a default dial plan that is not in