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Hey guys, I have my asterisk box running without a gui. I now need to monitor usage, calls, traffic of voice calls on this asterisk server. I cannot now install a gui because the configs will be wiped out, how can i go about monitoring all the above? -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype:

hi guys i have created a plugin for jquery for asterisk ajax interfacement. the interfacement work with ajam and on firefox work very well, the problem is with IE :-( an example: the url is: asterisk/mxml i want login on manager system and the string command is: action=login&username=myusername&secret=mysecret I have tested with firefoz and i receive the correct XML response, the

Hi, I just installed Asterisk 1.4.6. I didn't see http.conf in /etc/asterisk. Is there a seperate install for AJAM? I dug around a little and found only _one_ reference that refers to installation of AJAM: http://astrecipes.net/?n=217 In accordance with the instructions on this site I performed the following: svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui

Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMF&Channel=SIP/200-sdadsadkioah&Digit=1 the result is: <ajax-response> <response type='object' id='unknown'><generic response='Success' message='DTMF successfully queued' /></response>

I have setup an Ast 11.6 host and I want to login via AJAM. I setup manager.conf, http.conf described in the docs. When I login via the AMI it works fine (see below), but when I login via AJAM the same credentials fail (see further down) Can someone tell me how to fix this? ----------- Connection closed by foreign host. root at pbx:/tmp# telnet localhost 5038 Trying 127.0.0.1... Connected to

Hi I'm using asterisk 1.8.3.2 and have been implementing AJAM. I've noticed the final '>' is missing from every response I've had so far. Here is an example <ajax-response> <response type='object' id='unknown'><generic response='Success' message='Authentication accepted' /></response> </ajax-response Has anyone

Hi, We currently fire multiple HTTP requests (via multi-curl) to the AJAM interface in order to place calls. We are finding Asterisk has it's limits however, and I've found astmanproxy recommended for helping maintain the connections. This would prove particularly useful with multiple servers of course. However, in testing astmanproxy crashes with buffer overflows. This leads to the

I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-000000:append Cat-000000:newuser

Hi all I don't know what went wrong but i no longer seem to be able to compile asterisk. I first do : cd /usr/src/dahdi-linux-2.1.0.4 make clean ; make all ; make install cd /usr/src/dahdi-tools-2.1.0.2 ./configure ; make clean ; make all ; make install ; make config So far.... so good but then when i do : cd /usr/src/asterisk-1.6.0.6 make clean ; ./configure ; make all ; make install i

Hello there, I'm currently building a PHP-based software to help users make batch calls. Basically, users provide a script and list of phone numbers. The system calls those numbers and plays the script to whoever picks up the phone. Currently, the system does one call at a time via direct access to the Asterisk Manager Interface, but does not seem to be very efficient. Ideally,

Francis, I read in the list archives back that a future EventMachine release will support epoll on Linux (i.e., it''s in the trunk). Better still, is there a possibility that EM will rely on libevent so that it will be architecture independent (i.e. epoll on Linux, kqueue on FreeBSD/Mac OS X, /dev/poll on Solaris)? This is how memcached is implemented, and it would be helpful to be able

I'm looking for a graphical call manager for Asterisk. Ideally this would run locally on a desktop system or be web based. I'm familiar with iSymphony, FoP, HUDlite, AstAssistant, ADM, etc but I'm finding a common problem. They all let you transfer calls that connected to an 'extension' but not 'any' call on the system. For example, when a party calls into an * system,

Hi Ralph, Thanks for your suggestion. I have upgraded to Samba 4.1.14 (I know it's not the latest version but its the only version I could install on Gentoo at the moment). Unfortunately, my issues still persist. ----------------------------------------------------------------------- # smbclient -V Version 4.1.14 # smbtree -U AJames Enter AJames's password: WORKGROUP

Hi, i installed Cent Os 4.4 x64 (all packages )on my thinkpad . i have enough ram ie 2 gb and good processor Core 2 duo.2.5 ghz T9300 evrey thing seems to be running very slow. any help would be welcome thanx Sumit -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Everyone, I have tried over the past several days, unsuccessfully, to have my Linux client access folders and files shared under Mac OS X server. Mac OS X Info: Version 10.9.5 Username: AJames Hostname: bulbasaur Steps I took to share a directory: - System Preferences - Sharing - File Sharing - Add Shared Folders and let Everyone have Read Only access. - Under "Options"

I was wonderring if there is any quick way to check how many virtual registers are live-in at any give instruction of a function? Right now my understanding is that I would have to calculate the gen, kill, in and out sets and iterate over the entire functions CFG until the sets stop changing. Thanks _________________________________________________________________ Express your personality

Hey All - I've got an interesting problem, here is what I'm trying to accomplish: Six agents, two queues, three skill levels Queue A (queue B is the same) - Level 1 -- Agent 1 -- Agent 2 - Level 2 -- Agent 3 -- Agent 4 - Level 3 -- Agent 5 -- Agent 6 I'd like a call to come in to Queue A, ring agent 1 and agent 2 for X seconds, if agent 1 and agent 2 ignore the call, try

Dear R-help, First of all, thank you VERY much for any help you have time to offer. I greatly appreciate it. I would like to write a function that, given an arbitrary number of factors from a data frame, tabulates the number of occurrences of each unique combination of the factors. Cleary, this works: > table(horse,date,surface) <SNIP> , , surface = TURF

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

hi, i @asterisk - svn-ed asterisk from digium 1.6 - make install >> its running and i can access the CLI @gui then i -svned asterisk-gui from digium - installed - repointes apache /var/www/1234 >> /var/lib/asterisk/static_html >> now, i see the login box, but i dont have any credentials. tutorials are suggestion manager.conf, BUT I DONT HAVE that file, in fact /etc/asterisk is