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  • How to configure three IP address into single server

    - by user1363308
    I have Cisco device for call forwarding and three different system,I want to configure 15 and 16 server IP into 192.168.53.197 means eth0 --> 192.168.53.197 eth1 --> 192.168.16.15 eth2 --> 192.168.16.16 which work i have done with 15 and 16 individual , I will do some work on 197 after configuration eth1 and eth2. Means one system have three IP address but base IP address is 192.168.53.197

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  • How to receive the text which was sent by SendText

    - by thillai-selvan
    In asterisk I have sent a text message using SendText as follows I have two registered users in sip.conf file. sip.conf details [thillai] username=thillai secret=thillai host=dynamic type=friend allow=all context=test [selvan] username=selvan secret=selvan allow=all host=dynamic type=friend context=test Then I have created the necessary extensions like this extensions.conf file: [test] exten = 677,1,BackGround(thankyou) exten = 677,n,Dial(SIP/thillai) exten = 677,n,SendText('this is for testing') So when a caller is trying to call to extension 677 this text information will be sent. My question is how can I receive this text in the caller side? Any help will be much appreciated.

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  • SpeechBackground

    - by abinila
    Hai everyone, I have used the SpeechBackground application in asterisk. I used the version 1.6.0.6. I have a entry like, ;;SpeechCreate exten => s,1,SpeechCreate() exten => s,2,SpeechActivateGrammar(yesno) exten => s,3,SpeechStart() exten => s,4,SpeechBackground(demo-instruct) exten => s,5,SpeechDeactivateGrammar(yesno) I don't know which file I meed to give in SpeechBackground application. Please give me any idea. I have given the sound file from /sounds directory. If I call to 's' the call will be immediately released.I didn't get any audio sound. Please any one help me...

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  • Using flash for videoconferencing

    - by PHP
    Hi I'd like to set up a videoconferencing (1-1 initially) system using Flash as the client. I've tried some web searches but couldn't find much; any pointers on the following would be great: Does the standard browser Flash 10 plugin contain everything needed for P2P video conferencing, or is server support needed? Does anyone have any experience of opensource server videoconferencing products to be used with Flash on the browser side? E.g. would Asterisk work? Something like http://www.red5chat.com/ ? (ideally I'd like to be able to record videoconferences also). Any licencing issues? Many thanks

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  • Centreon setup (web part) config not writeable

    - by Evgeniy
    I have a problem with Centreon installation. At end of install process, I stopped at 10th step: Writable Centreon Configuration File (centreon.conf.php) Critical: Not Writeable /etc/centreon/ asterisk:asterisk (755) Should be : (755) Generate Centreon configuration file OK /etc/centreon/centreon.conf.php asterisk:asterisk (755) Should be : (755) Generate Centstorage configuration file OK /etc/centreon//conf.pm asterisk:asterisk (755) Should be : (755) But permissions and owner are OK. And I tried chmod -R 777 /etc/centreon Please help to understand what's wrong. UPD: Screenshot: http://gyazo.com/82701d119a5e0d98947ad35406a5b8f3

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  • How to use regex to match ASTERISK in awk

    - by Ken Chen
    I'm stil pretty new to regular expression and just started learning to use awk. What I am trying to accomplish is writing a ksh script to read-in lines from text, and and for every lines that match the following: *RECORD 0000001 [some_serial_#] to replace $2 (i.e. 000001) with a different number. So essentially the script read in batch record dump, and replace the record number with date+record#, and write to separate file. So this is what I'm thinking the format should be: awk 'match($0,"/*FTR")!=0{$2="$DATE-n++"; print $0} match($0,"/*FTR")==0{print $0}' $BATCH > $OUTPUT but obviously "/*FTR" is not going to work, and I'm not sure if changing $2 and then write the whole line is the correct way to do this. So I am in need of some serious enlightenment.

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  • not able to register sip user on red5server, using red5phone

    - by sunil221
    I start the red5, and then i start red5phone i try to register sip user , details i provide are username = 999999 password = **** ip = asteriskserverip and i got --- Registering contact -- sip:[email protected]:5072 the right contact could be --- sip :99999@asteriskserverip this is the log: SipUserAgent - listen -> Init... Red5SIP register [SIPUser] register RegisterAgent: Registering contact <sip:[email protected]:5072> (it expires in 3600 secs) RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout RegisterAgent: Failed Registration stop try. Red5SIP Client leaving app 1 Red5SIP Client closing client 35C1B495-E084-1651-0C40-559437CAC7E1 Release ports: sip port 5072 audio port 3002 Release port number:5072 Release port number:3002 [SIPUser] close1 [SIPUser] hangup [SIPUser] closeStreams RTMPUser stopStream [SIPUser] unregister RegisterAgent: Unregistering contact <sip:[email protected]:5072> SipUserAgent - hangup -> Init... SipUserAgent - closeMediaApplication -> Init... [SIPUser] provider.halt RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout please let me know if i am doing anything wrong. regards Sunil

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  • Is it possible to add asterisk serveralias to virtualhost without modifying httpd.conf manually?

    - by Favourite Chigozie Onwuemene
    Is it possible to add wildcard serveralias (example: *.somesite.com) in an apache server without modifying httpd.conf manually? I use a DNS different from my hosting server and i have added asterisk A record to my DNS to point all request like (test.somesite.com,test2.somesite.com) to my hosting servers IP, but i don't see anyway of adding asterisk serveraliases to apache httpd.conf file in my cpanel. Pls is there a solution?

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  • Stop fail2ban stop/start notifications

    - by michael
    If the server is restarted, or even if fail2ban is stopped/start it sends a notification. [asterisk-iptables] enabled = true filter = asterisk action = iptables-allports[name=ASTERISK, protocol=all] sendmail-whois[name=ASTERISK, [email protected], [email protected]] logpath = /var/log/asterisk/messages maxretry = 5 bantime = 259200 Removing the sendmail-whois stops it, but it also stops the ban notifications, how can I get it to stop notifying me when the process starts/stops? Thanks

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  • Declaring pointers; asterisk on the left or right of the space between the type and name?

    - by GenTiradentes
    I've seen mixed versions of this in a lot of code. (This applies to C and C++, by the way.) People seem to declare pointers in one of two ways, and I have no idea which one is correct, of if it even matters. The first way it to put the asterisk adjacent the type name, like so: someType* somePtr; The second way is to put the asterisk adjacent the name of the variable, like so: someType *somePtr; This has been driving me nuts for some time now. Is there any standard way of declaring pointers? Does it even matter how pointers are declared? I've used both declarations before, and I know that the compiler doesn't care which way it is. However, the fact that I've seen pointers declared in two different ways leads me to believe that there's a reason behind it. I'm curious if either method is more readable or logical in some way that I'm missing.

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  • Strange. Asterisk key plays random Windows sound when pressed

    - by Charles
    This is a new one on me. When I press the "asterisk", or * button on my number pad (but not SHIFT+8), Windows makes either a "Exclamation" or "Windows ding" sound. I haven't noticed a pattern to which sound is made. Logitech K200 keyboard No special key mapping software or Logitech software running Realtek sound to stereo through optical cable. Visual Studio 2010, Chrome, Fiddler, WinRAR, Notepad++, and Dropbox running. No unusual behavior otherwise. A solution isn't terribly important but my curiosity is both piqued and stumped. This doesn't normally happen and nothing odd has taken place otherwise. Ideas?

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  • Sniff UNIX domain socket

    - by gonvaled
    I know that some process is writing to a certain unix domain socket (/var/run/asterisk/asterisk.ctl), but I do not known the pid of the sender. How can I find out who is writing to the socket? I have tried with: sudo lsof /var/run/asterisk/asterisk.ctl but it just list the owner of the socket. I would like to know who is writing / reading to this socket, and I would also like to sniff the data. Is this possible?

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  • Pinning based on origin of a reprepro repository.

    - by Shtééf
    I'm on Ubuntu 10.04, and trying to set up a repository using reprepro. I'd also like the pin everything in that repository to be preferred over anything else, even if packages are older versions. (It will only contain a select set of packages.) However, I cannot seem to get the pinning to work, and believe it has something to do with the repository side of things, rather than the apt configuration on the client. I've taken the following steps to set up my repository Installed a web server (my personal choice here is Cherokee), Created the directory /var/www/apt/, Created the file conf/distributions, like so: Origin: Shteef Label: Shteef Suite: lucid Version: 10.04 Codename: lucid Architectures: i386 amd64 source Components: main Description: My personal repository Ran reprepro export from the /var/www/apt/ directory. Now on any other machine, I can add this (empty) repository over HTTP to my /etc/apt/sources.list, and run apt-get update without any errors: Ign http://archive.lan lucid Release.gpg Ign http://archive.lan/apt/ lucid/main Translation-en_US Get:1 http://archive.lan lucid Release [2,244B] Ign http://archive.lan lucid/main Packages Ign http://archive.lan lucid/main Sources Ign http://archive.lan lucid/main Packages Ign http://archive.lan lucid/main Sources Hit http://archive.lan lucid/main Packages Hit http://archive.lan lucid/main Sources In my case, now I want to use an old version of Asterisk, namely Asterisk 1.4. I rebuilt the asterisk-1:1.4.21.2~dfsg-3ubuntu2.1 package from Ubuntu 9.04 (with some small changes to fix dependencies) and uploaded it to my repository. At this point I can see the new package in aptitude, but it naturally prefers the newer Asterisk 1.6 currently in the Ubuntu 10.04 repositories. To try and fix that, I have created /etc/apt/preferences.d/personal like so: Package: * Pin: release o=Shteef Pin-Priority: 1000 But when I try to install the asterisk package, it will still prefer the 1.6 version over my own 1.4 version. This is what apt-cache policy asterisk shows: asterisk: Installed: (none) Candidate: 1:1.6.2.5-0ubuntu1 Version table: 1:1.6.2.5-0ubuntu1 0 500 http://nl.archive.ubuntu.com/ubuntu/ lucid/universe Packages 1:1.4.21.2~dfsg-3ubuntu2.1shteef1 0 500 http://archive.lan/apt/ lucid/main Packages Clearly, it is not picking up my pin. In fact, when I run just apt-cache policy, I get the following: Package files: 100 /var/lib/dpkg/status release a=now 500 http://archive.lan/apt/ lucid/main Packages origin archive.lan 500 http://security.ubuntu.com/ubuntu/ lucid-security/multiverse Packages release v=10.04,o=Ubuntu,a=lucid-security,n=lucid,l=Ubuntu,c=multiverse origin security.ubuntu.com [...] Unlike Ubuntu's repository, apt doesn't seem to pick up a release-line at all for my own repository. I'm suspecting this is the cause why I can't pin on release o=Shteef in my preferences file. But I can't find any noticable difference between my repository's Release files and Ubuntu's that would cause this. Is there a step I've missed or mistake I've made in setting up my repository?

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  • Why does 'sort' ignore special characters, like the asterisk?

    - by Aaron Digulla
    I thought that sort would sort common prefixes together but that doesn't always happen. Take this input for example: AT0S*eightieths AT0S*eyetooth's AT*ad AT*Ad AT*AD AT*Eydie AT*eyed ATF*adv ATF*ATV ATF*edify ATF*Ediva ATFKT*advocate ATFKTNK*advocating ATFKT*outfought ATFKTS*advocates ATHT*whitehead ATHT*Whitehead AT*id AT*I'd AT*Ito AT*IUD ATJ*adage ATNXNS*attention's ATNXNS*attenuation's ATNXNS*autoignition's AT*oat AT*OD AT*outweigh AT*owed ATP0K*idiopathic ATP*adobe ATT*wighted ATT*witted ATT*wooded AT*UT AT*Uta AT*wowed AT*Wyatt ATX*atishoo After sort, I'd expect all the AT* to end up in one chunk but when you run this data through sort, the output == input. Why is that? I'm not specifying any option to ignore non-alphabetic characters or anything. Just sort dict > out. My version of sort comes from coreutils 8.5-1ubuntu3.

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  • License for Opensource project

    - by asterisk
    I am newbie in the open source community world. I am planning to develop a open source project, hosted on github. The project would be using other open source components like- NHibernate, FNH, Log4net, CommonLibrary, Autofac, Quartz.Net Scheduler etc etc My questions are: Would there be any restrictions on using above OSS components? for example: I plan to use MIT license, but Quartz.Net Scheduler uses Apache license, would there be any restrictions? How do I get a license for my own project? Do I need to register my project somewhere? What is the best practice to mention credits to the OSS compoenents used? Many thanks,

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  • C# How do I replace an actual asterisk character (*) in a Regex expression?

    - by fraXis
    Hello, I have a statement: I have a string such as content = "* test *" I want to search and replace it with so when I am done the string contains this: content = "(*) test (*)" My code is: content = Regex.Replace(content, "*", "(*)"); But this causes an error in C# because it thinks that the * is part of the Regular Expressions Syntax. How can I modify this code so it changes all asterisks in my string to (*) instead without causing a runtime error? Thanks.

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  • How to play WAV file through Network Paging Interface

    - by BGM
    In our building we have a Viking Paging ZPI-4 Interface for our intercom. The interface receives data from our Asterisk Phone system via a Cisco SPA112 Port Adapter which has it's own IP address on the network and converts digital into analog. Asterisk plays the "5" tone and then allows the user's voice to commence over the connection. Now, what I want to do is to play a wav file over this Viking Paging device using the Cisco Port Adapter. I know how to get Asterisk to do it, but I want to do this without Asterisk. I want some kind of program that can talk to the Cisco Port Adapter and then transmit the wav file into the Viking Paging Device. What kind of program do I need to get or make? Now, I found this link if it helps anyone with ideas. I also found this information, but I'm not sure how to apply it. I also found this, but it involves an arduino. However, I already have the analog-to-digital convertor, and the Viking will handle sending sound over the paging speakers. I just need to know how to send the wav file to the Viking via the Port Adapter. So far, I know my wav file should be formatted as 8bit mono, and I need to send the "5" tone to open the Viking Pager's channel. [update] I am trying to figure out if I can use VLC player to stream to the ipaddress of the Port Adapter. So far I'm not having success with that, and don't even know if it will work. Windows Media Player has a streaming option too. I am thinking that since the Cisco Port Adapter thinks it is a sort of phone, that the only way this can be done is via SIP.

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  • replace only one variable in web.py templates

    - by Asterisk
    I am passing variable to template in web.py and have the same condition in some places. Like this: $if myvar=="string1": $passed argument1 ............ $if myvar =="striung2": $passed argument2 If say myvar is "string1" and I pass passed = "AAA" then I have AAA argument1 on my page, but the other if statements get replaced by empty string? How to avoid that? I.e. how to leave other statements intact?

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  • Turn old rack server into a telephone server

    - by Jake Elsley
    I have an old server lying around and I am thinking of using it as an internal telephone server. Its main use would be to set up a 1 to 1 telephone system that could be used internally to connect to different users in different offices. I have looked at software like Asterisk, but it seems that this is mainly for external telephone systems. Is this possible to do with Asterisk (or other software) or is this not possible without involving a VOIP company?

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