similar to: how to get /var/run/asteris/asterisk.ctl

Have a nice day, Scott Berry E-mail: N7zib at northlc.com I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. I am however having some problem where I cannot do "asterisk -r" and hook up to the asterisk CLI. I have checked to see that

Hello there, I am wondering if some one could tell me on the average in the U.S. What does a person with DCap certification make on a standard Asterisk installation and configuration process as well as a Sip Master. I am looking to go to the Asterisk course and I am blind and have a state agency possibly paying for my training and I would like to find out what the average wages are so that I can

Hello there, I am reading Asterisk: The Future of Telephony Chapter four. I am using a Ubuntu box with Asterisk precompiled at this time so I can learn. I am finding that I am having a problem when I do "asterisk -r" from the command line. It says: Unable to connect remotely (are you sure that /var/run/asterisk/asterisk.ctl is available.) The answer to this question is yes. I also

Hello there, I am trying to get Asterisk set up by using the book Asterisk: The Future of Telephony. I am on Chapter 4. I have have set up Zaptel and zapata.conf and also set up extensions.conf and when I run "asterisk -r" at the Gnome-terminal to connect with Asterisk I get the following message: Unable to connect with remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It

Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan

I'm trying to get a script working on a website to send out automatic email reminders to customers reminding them monthly to change furnace filters. I haven't got one running successfully, yet. That made me think - could it be done with a phone call using Asterisk? A monthly automated phone call to remind people to change their furnace filter? I have no ability to figure this out

Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus

Is there any script guru on the list that can help me. I'm trying to start asterisk with nice -5. Normally the command would be: nice -5 asterisk but asterisk start from the scrip on Gentoo as -U asterisk -G asterisk Here is the script: =============== depend() { need net use zaptel } start() { local OPTS USER GROUP if [[ -n "${ASTERISK_NICE}" ]];

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

I'm trying to make H.323 trunk between AVAYA&Asterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob.

Hi, What is today's status of Asterisk connections management ? Is Astmanproxy still recommended or shall something else be used ? Astmanproxy works for me with Asterisk 1.4 but it seems this software is not updated recently (recent patches are not merged, last modification dates from 16 months). Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone tried the Winsip sotware to test Asterisk? _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

>The old "lock-in" model of the telecom world was nasty and monopolistic and >expensive, but by jove it just worked. >Because as you've seen from this list ANYTHING and EVERYTHING can, and >does, go wrong with this technology. Yes, true, but this is the nature of the beast. It (Asterisk) is an infinitely configurable platform that runs on anything. This essentially

To manage Asterisk users, I''ve made a define that can be used like this: asterisk::user { "sip.login": fullname => "Sip Extension 223", secret => "allyourbase", extension => "223", } I''m trying to put user definitions in a single string, and make something like the following. Should I go down this way, or should I use

Hello I fixed my problem. I changed user and group in /etc/mISDN.conf: <devnode user="asterisk" group="asterisk" mode="644">mISDN</devnode> now it works again! Thanx 4 help! ;) Ingrid -- "Bonnie & Clyde der Postmaster-Szene!" approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org --------------

Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc:

CentOS 4.1, Xen 2.0.7. My dom0 is booting up but can''t get to the outside network because eth0 won''t start due to, apparently, Xen providing a different mac address than expected. How do I tell Xen to start eth0 at the bootup with a specific mac address that lines up with the HWADDR= line in my ifcfg-eth0? Note that a normal bootup into CentOS 4.1 outside of the Grub Xen bootup

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

Hi all, we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances behind FW. Problem we face is that when we reboot the DC Asterisks, the trunks (SIP or IAX) become alive from DC Asterisks to clients ones but UNAVAILABLE the other way. In clients logs we see Registration for 'XXX at

What kinds of problems were you having? I'm on 1.4.0 and chan_gtalk.so simply doesn't load. Of the 146 files in the /usr/lib/asterisk/modules/ directory, asterisk loads 144 of them, omitting only chan_gtalk.so and res_jabber.so. Connected to Asterisk 1.4.1 currently running on monkey (pid = 9371) Verbosity is at least 3 foo*CLI> module load chan_gtalk.so [Mar 7 10:23:07]