similar to: Asterisk PBX -> RT Integration

Does anyone know if this unit works with Asterisk? Thx. B.

Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 <..> Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware

Hello list! I?m working on a fresh Asterisk install over CentOS7 base. I?m using ?Asterisk. The Definite guide? book as a reference. I connect and work using SSH Problem I have - I can?t connect to asterisk from remote. Getting error: $ sudo asterisk -rvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Yes, it exist, and service runs: [asteriskpbx at

Hello list, I'm hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). It's my first Asterisk installation and meant for simple home use. I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently it's configured for Ekiga so I can test. In a few weeks I'll change to a Telco SIP

You can try to run "make" in the linux source-folder. I had the same problem Running FedCore 4 on a Dual Xeon Server and running make fixed the error ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Teesdale Sent: 15 February 2006 12:19 To: asterisk-users@lists.digium.com Subject:

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

[wrapping disabled to allow for easier review] Yet another SIP authentication problem. I have SER running, and passing calls to a PRI-enabled Asterisk server from a large range of Media Terminal Adapters, and a few other Asterisk systems set up as "clients". I have this PRI-enabled Asterisk server functioning as a very simple media gateway to hand off my toll-free calls to a PRI -

Is there a digium twitter 'user' to follow that only tweets important announcements and release information? If there is not, I think there should be. It would be highly utilitarian to get an SMS when there is an update to Asterisk, Dahdi, ADA etc, but I don't want to be bothered real-time with asteriskpbx tweets like: "Anyone trying anything cool with Asterisk over the

Anyone in New Zealand using AsteriskPBX? If so, what hardware are you using to connection to Telecom's lines? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030619/d5c12ca0/attachment.htm

Hi, In digium website. http://store.yahoo.com/asteriskpbx/newitastdmde.html It is said Dev Kit PCI card works with 2.4 kernel. I am wondering if it is also working with 2.6 kernel? Anyone knows? Thanks Michael

Hi there, I know this question is kind of stupid. But, I don't know anywhere else to ask. I've received some answers when I asked about the need of having a zaptel interface to make the meetme application work, that said that it was better to have a real hardware then the zaptelrtc software modules. So, my question is, would any of the following cards work with the meetme

HI all: I have spent a large amount of time configuring/installing phones connected to Asterisk. Halfway through the process I discovered that my Cisco7960 with 2 7914 expansions was not supported in the SIP protocol. After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of configuring SCCP to properly work with Asterisk. So far I have gotten the phone to dial and receive calls

On Wed, Jun 21, 2023 at 05:19:11PM +0000, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls using the trunk are rejected with a 403. Using pjsip >

Hi, I am an experienced developer with Windows and familiar with Linux. I am looking for a SIP solution. 1) How does Asterisk compare to VOCAL in terms of support. 2) Is Asterisk free? 3) Where are the docs? Or even better. Where do I start? 4) Will it run on RH9? Thanks in advance. Costas -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

The hardware we are planning to use is: Micronet SP5050 FXO Gateway http://www.micronet.com.tw/Products/VoIP/SP5050.asp Micronet SP5100 IP Phone http://www.micronet.com.tw/Products/VoIP/SP5100.asp We are hoping to use this hardware along with AsteriskPBX to replace our aging PBX system. What I want to acheive is: * Any incoming call from PSTN (via gateway) rings on the receptionists phone for

I've been searching the Internet for information regarding the replacement of astdb with a modern sql engine. There are several reasons one would like to do this. First of all, external applications have a hard time reading/writing to the now-old astdb format. Also (and this is what interests me most), the sql astdb could easily be clustered throughout several servers (I'm looking for a

Hello, I acquired a Diva Server 4BRI for an AsteriskPBX system and installed a 2.4.20 kernel modified with the latest eicon capi2.0 support from www.melware.net. Capi support seems to work fine: Asterisk+chan_capi answers the calls. However if I try to use my old isdnutils apps by loading a i4l compatibility layer then I have the following problems: - if I use the diva2i4l module and start the