similar to: Missing Config Files under /etc/asterisk

Displaying 20 results from an estimated 2000 matches similar to: "Missing Config Files under /etc/asterisk"

2011 Apr 20
2
issue with installtion asterisk
hello all, I have installed centos 5.5 ( linux text) and I have updated it with # yum install bison bison-devel================?ok # yum install ncurses ncurses-devel==========?ok # yum install zlib zlib-devel===============?ok # yum install openssl openssl-deve=======?ok # yum install gnutls-devel============ ==?ok # yum install gcc gcc-c++============?ok # yum install newt
2011 Jul 01
1
Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted
Hi Please help me understand about the below issue ? [root at asterisk1 ~]# /etc/init.d/asterisk restart Stopping safe_asterisk: [ OK ] Shutting down asterisk: [ OK ] Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted
2010 Mar 24
6
Restarting Asterisk using a script - Thanks to all -
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
2011 May 24
3
How to enable the addon in the Asterisk 1.8 compilation
Hi All; To enable the compilation for the addon that is coming with Asterisk 1.8 when doing compilation for the Asterisk, what should I do? Regards Bilal
2011 Feb 23
4
secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1
Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss
2010 Oct 27
1
Extension notation in default ViciDial installation
Hello List, A few days ago I installed ViciDial on a server, and while looking to the default 'extensions.conf' file, I saw this line: exten => _010*010*010*015*.,1,Dial(${TRUNKTESTast}/${EXTEN:16},55,oT) Can someone point me out to the Asterisk documentation part where explains how to use server IP's as extension number? I could not see it in the ATFOT2 book, and I would
2011 May 19
3
Manager logged on/off messages
Hi Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
2011 Mar 10
2
Is H323 supported when installing Asterisk from Digium Yum repository?
Hi everyone, Installed asterisk from yum repository but I think H.323 is not supported as I tried commands like this and they don't work: - *h.323 debug*: Enable chan_h323 debug - *h.323 gk cycle*: Manually re-register with the Gatekeper - *h.323 hangup*: Manually try to hang up a call - *h.323 no debug*: Disable chan_h323 debug - *h.323 no trace*: Disable H.323 Stack Tracing
2009 Jun 10
1
Dialer program
Hello, I am looking for a dialer program, free or not, that allows me to perform scheduled calls, generate reports and let me upload sound files. Is there something that fits these features?. If there is not any product like I mentioned before I am interested to build this kind of software but I need ideas to make it useful for technical and non-technical people. I don't want to spend my
2009 Oct 28
1
Clear pending SIP channels
Hi all, I have a question regarding pending (zombie) SIP sessions: on Asterisk CLI, with command 'sip show channels' , I see two channels in use with callID and other infos detailed; also 'sip show inuse' give me same result (in terms of channels usage): Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message xx.xx.xx.79 209
2010 Aug 30
1
Asterisk routing to SoftSwitch
Dear All, First, I am not so much experienced in Asterisk. I need asterisk to route the call to soft switch when the caller is not in its extensions list. And also when routing to soft switch, a number 4327 has to be added in the caller's number and then routed. I think its not so hard in asterisk. Please help me. Regards, Pratik -------------- next part -------------- An HTML attachment
2011 Jan 19
1
intermittent problem on 1.4
We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline->voipfone->asterisk 1.4->voipfone->UK landline About 1 in 3 times the call at the final landline is silent and we see "RTP Read too short" scrolling on the console log. Where do we
2011 Apr 27
1
h323 with NAT
Hi list, I've been beating my head for about 3 days on this one. I have Asterisk 1.4.41 installed using openh323. As long as I'm inside my firewall, everything is hunky-dory. When I move to server on another subnet, I'm still able to connect, but no longer have sound. Any good pointers or suggestions? Thanks Danny Nicholas
2010 Nov 19
2
Installing Asterisk to it's own directory
I'd like to start playing with 1.8, however I don't want to potentially damage anything on my existing 1.6.2 install on my production server. I'd like to test 1.8 against my existing configs leaving my 1.6.2 install untouched. Looking at the output of ./configure --help suggests that it's possible to install Asterisk into another prefix of my choosing, but as this is
2011 Jun 08
1
After wiki.asterisk.org was upgraded my user no loger exists.
Hello Guys, After the Wiki was updated to the 3.5.X version, my username is no loger available: user: khratos mail: jpe at slackware-es.com I had some documents on my personal space. Is there a way to recover the account? Regards, -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs
2009 Feb 07
3
VPN and Asterisk
One of my user was asking, can he use VPN to access asterisk ? What does it mean ? And its possible ? How ?VPN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090207/325c0670/attachment.htm
2010 Sep 22
5
http://www.asterisk.org/downloads naming schema
Hi! Since some time the download of the newest Asterisk does not contains the version number anymore, but is just called "asterisk-1.4-current.tar.gz" This gives me a tarball where I do not know the version without looking into the tarball. Thus, IMO it would be very useful to switch back to old schema war the download contained the version number. Thanks Klaus
2010 Dec 20
3
cdr_mysql stopped working
I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql table for CDR's today there are no entries since the update. I have rebuilt and re-installed and re-started asterisk still no CDR's flowing to mysql. I did not change any configs. I checked to make sure that the cdr_mysql option was selected under the make menu options. The module shows it is there when I do a
2009 Feb 03
2
Broken Pipe error while using UpdateConfig command
Hello List, I have been working on a little PHP software that uses AMI's UpdateConfig command in order to modify some of it's config files. I was working with 'Asterisk 1.4.22.1' and everything was working. After upgrading to 'Asterisk 1.4.23.1' I receive a lot of errors of the type: ERROR[11505]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
2010 Sep 10
7
A way to check against a list of numbers?
Does anyone have a suggestion on how to handle this? For example, if I have a list of numbers that I want to go out a certain sip channel and another that I want to go out the dahdi device, is there a way to do this? None of the numbers will fit into a pattern, so just plain pattern matching won't do. The most straightforward way would be to just define explicit patterns. Obviously that