similar to: Login by AMI ok, by AJAM fails

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 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 all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

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

this purpose. I have been reading the voip-info pages and have set up AJAM and can get results from doing http requests to the asterisk server, however, this is in the form of an action, such as login, rather than subscribing to an event. I have been looking here for information http://www.voip-info.org/wiki/view/asterisk+manager+events Does anyone know if subscriptions is possible with AJAM?

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>

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,

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

Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf file: [3002] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Allow codecs in order of preference allow=ilbc

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 have two Asterisks identically installed on two computers. One of them is directly connected to the Internet, the other one through a NAT router (Netgear MR314). On the one behind the router I have an X100P card installed for PSTN connections. In the local LAN of each PBX they are several hardware IP phones (Cisco 7960 and 7940 with SIP images, firmware image P0S3-04-4-00.bin). I have

The Asterisk Development Team would like to announce the release of Asterisk 15.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Release Announcements --------------------- This is the latest stable release of the Samba 4.19 release series. Changes since 4.19.4 -------------------- o? Ralph Boehme <slow at samba.org> ?? * BUG 13688: Windows 2016 fails to restore previous version of a file from a ???? shadow_copy2 snapshot. ?? * BUG 15549: Symlinks on AIX are broken in 4.19 (and a few version before ????

Release Announcements --------------------- This is the latest stable release of the Samba 4.19 release series. Changes since 4.19.4 -------------------- o? Ralph Boehme <slow at samba.org> ?? * BUG 13688: Windows 2016 fails to restore previous version of a file from a ???? shadow_copy2 snapshot. ?? * BUG 15549: Symlinks on AIX are broken in 4.19 (and a few version before ????

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"

On Tue, Jan 13, 2015 at 10:41:36PM +0000, Am Jam wrote: > 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

voipstunt: "Forbidden - wrong password on authentication for INVITE to ...." I have paid, the password was not changed, ... I have no idea why. Is there anything what I can do to get this "failed" call over to another provider, so that the user can complete the call? (Dialstatus was an idea, but the line does not show up in CLI) [Apr 5 09:22:36] -- Executing

Ok, to review, I'm trying to get Avaya 9608G to come up in a pure Asterisk environment-- no Avaya SBC or gateway or any other Avaya gear in sight. I have the phone working to the point where it boots up properly, then displays a Username and Password prompt, and says its extension is 123 and the time is 4:57p, which is wrong. But please don't tell me the only way to program up each