similar to: [asterisk-dev] Question

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

> > Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

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

Hi, The problem solved After installing new zaptel drivers, we ran the "genzaptel" command to generate /etc/zaptel.conf file,checked with "zttool" command and the card status was "Yellow alarm/Blue alarm/Recovering" and the card LED was blinking red and green. The problem was with the generated zaptel configuration., but not with the pin

=20 Andrew Fenn wrote: > You don't need their program to use justvoip, voipdiscount, etc=2E You > can use any sip client to connect to Betamax servers=2E Try Twinkle=2E >=20 > On Mon, Jul 27, 2009 at 11:24 PM, miroa84<wineforum-user at winehq=2Eorg> wrote: >=20 > > I tried to install justvoip several times and I cannot install it=2E Can somebody tell me how to

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> >

Hi I put a snapshot of our current packages updates.xorcom.com . They are available from the deb source deb http://updates.xorcom.com/test sarge main (this is s/rapid/test/ of the name of the source of the stable version) Changes include: * Fixed and simplified zaptel detection * Support for spandsp: compiled but not yet tested * IAX extensions * Mail server configuration * The script

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

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

Hi, I am on amd64 Linux and not really too happy with twinkle, linphone and ekiga. Unfortunately, X-Lite and Zoiper, even though they provide Linux versions (w00t!) have only x86 versions for download. Do you guys know of amd64 versions of those, or can you recommend other softphones that will run on amd64, or which come with source code? Thanks, -- martin | http://madduck.net/ |

On Thu, Jul 15, 2010 at 10:19:10AM -0700, Steve Casto wrote: > > https://issues.asterisk.org/view.php?id=17529 > Thanks Tzafrir: > Unclear on how to apply patch, here is what I get: > [root at localhost asterisk-1.4.32]# patch -p1 < ../bug17529.diff.txt > can't find file to patch at input line 5 > Perhaps you used the wrong -p or --strip option? > The text

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

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

Hi folks I've tried asking this in private mail for quite some time, but sadly got no reply. So I end up needing to raise the question here. Sangoma's s setup process includes a small patch to Zaptel. I have some technical reservations with htat patch. It seems that under certain circumstances it may cause exexpected behaviour when used with other Zaptel channel drivers. I also don't

Hi Debian has a package for chan_oh323 (the original, external h323). It is not maintaind for quite some time AFAIK and also AFAIK offers no real atvantages over chan_h323. So I'd like to remove it. Before I do that, I have some questions, as I'm not familiar with H.323 channels: 1. Are there any useful features oh323 supports that h323 doesn't? That the version of h323 in 1.4.21

While I read on some other mailing list that the human ear is a poor testing device, it is still a widely available testing device and I often don't have anything better. In order to help that device better detect sound quality issues, I tend to prefer to use lengthy music files. Once I'm familiar enough with the music I can sense "something is wrong" with relatively little