similar to: asterisk services not starting up

here is the issue phones freepbx-2.7.0]# ./start_asterisk start STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. mpg123: no process killed ----------------------------------------------------- Asterisk could not start! Use 'tail

Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at

Using centos 5.2, I want to use a kickstart file to select packages in order to have an unattended install onto a bare-metal server. Is there any way of finding out which packages I need to compile, build and run asterisk ? I generally want to build all modules in asterisk and the wct4xxp zaptel module I want to be able to not select any groups in the kickstart file, but only select

Dear All Can you please do me favor and let me know how can I stop my Asterisk ? Can you please confirm if the following procedure is correct to stop it ? #/etc/init.d/asterisk stop #cd /etc/init.d #chmod 0000 asterisk Let me thank you in advance -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at your install and they said we are having a NAT problem but didn'ttell me if it was with the asterisk conf or the Cisco ASA. _________________________________________________________________ Hotmail: Trusted email with Microsoft?s powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/

Hoping someone can help me understand what is happening here; we start asterisk as a service at boot (actually, with heartbeat) on CentOS using the asterisk init script installed with "make config" upon reboot of the server (when the asterisk service is first started by heartbeat) we get color in the console when we connect to it using asterisk -r after the execution of

hi, i have solaris 9, the kind that runs on a pc. i tried downloading gcc-3.3.2-sol9-intel-local.gz; gcc-3.4.6-sol9-x86-local.gz and gcc_small-3.3.2-sol9-intel-local.gz. none of them decompressed properly. i tried it a few times. i downloaded from "sunfreeware.com". any suggestions? thanks. much appreciated. givon

dear allI have a data set with three types (Tree, Sapling, Seedling). I have estimated the correlation values. now i need to bring all the correlation values in a table like the one i have shown in attached file with R codes.could you please give me idea on this problem thanking you MSNepal _________________________________________________________________ Hotmail: Trusted email with

hi all, can anyone give me a guide on any asterisk admin solution / interface for config management, and monitoring? No database use is intended and I prefer open source. Thanks for support. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160516/98f6e448/attachment.html>

I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"

here is the debug from the CLI. I think I know where the problem is I just can figure out how to fix it. The IP in the From and To i think is where the problem is. When I make an outbound call. i get the message "the call cannot be completed as dialed". if i call another ext it works. I posted the debug for both calls. ==============outbound call=========================== <---

I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. So, other than a bit of disk space, is there any reason why I shouldn't install KDE when I set it up ? Is there any great disadvantage to running the server in init level 5 (ie KDE, xorg,

We have a problem in that calls are dropped after 15 minutes (on both internal and out going calls, incoming calls do not seem to have that limit) How do we fix it? This is the version on that PBX Kernel Linux(x86_64)-2.6.18-371.1.2.el5 Elastix elastix-2.4.0-8 elastix-a2billing-1.9.4-5 elastix-addons-2.4.0-10 elastix-agenda-2.4.0-14 elastix-asterisk-sounds-1.2.3-1

In an earlier thread it was mentioned I could use postfix stop to stop postfix. I'm trying to get sshd started and starting on boot. I did chkconfig sshd on and that worked fine, but then tried sshd start, but that didn't work. It looks like I need to do service sshd start (I did just that and it is now started). Why the difference? Thanks, -wes

Hello, Reading through the Wiki: "Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony applications using Asterisk" My question is, what is the benefit of using Lua? I recently

I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? -- Thanks in advance,

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi All, I have installed astriesk 6 and am able to make calls using sip x-lite. Its working as I expected. Now I want to make call from sipx-lite to PSTN using asterisk. can any please suggest me which Hardware card that I can buy? and use ( pl give me all ths list of cards.... which are good.). 2) what is that I need to do after buying the card to make it talk to the real world PSTN network?

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

On Fri, 18 Mar 2016, Trey Hilyard wrote: > I thought this would be as easy as > exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST