similar to: AMP or Asterisk

Hi all, I have been using Asterisk for a while now, and loving it. Just about to update to 1.0 (running like 0.93) I was wondering if anyone has any expertise in the implementation of AMP onto an existing Asterisk install? The instructions for it all deal with a fresh install of Asterisk, and I would hate to be forced to re-configure. Any advise would be greatly appreciated. David A. Morrow

Hi all. Please forgive me if this question is of the "RTFM" variety. I am new to Rsync and just attempting to determine if it will meet my needs with respect to one problem I am attempting to solve. I have a web application which is running on Apache2/PHP which allows users to upload files. I would like to add a redundant server for load-balancing and failover purposes but need to

Hi all, I have Winbind authentication up and running properly (thanks to new, easy to use features of Redhat Ent 4). My question is this. I know that I can, by massaging /etc/pam.d files manually, have Winbind/Samba automatically create a home directory for each user that logs in, but I am wondering if Samba/Winbind can instead map to their home directory as defined in their Windows profile

Hi all, I have asterisk installed and working just fine with a couple of Cisco IP Phones. I am now ready to pilot connectivity to PSTN and am wondering what hardware would be recommended to make minimum connectivity to the public telephone network. I am think ISDN as I would like a few external lines to be accessible. David A. Morrow Technical Systems Lead Autodata Solutions Company

Hi all. I am trying to find out what the most popular soft phone for Windows is for use with Asterisk. SIP or IAX? David Morrow Technical Systems Lead Autodata Solutions Company David.Morrow@Autodata.net http://www.autodatasolutions.com <http://www.autodatasolutions.com/> Tel: (519) 963-3020 Fax: (519) 451-6615 < Lead, follow or get out of the way! > This message has

Hi all. I hope this is not a RTFM-type question, but I've been unable to find a searchable archive of this mailing list....... I recently began investigating using Dovecot/Postfix/MySQL solution. I've been following the documentation http://wiki.dovecot.org/HowTo/DovecotLDAPostfixAdminMySQL?highlight=%28m ysql%29 While everything seems to have gone right, I have been unable to

Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and Dial(SIP/100,,wW) Specifically, what is the |20|Ttr ? I cannot seem to find any reference which would indicate

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten =>

Hi all, I was wondering if someone could lend a little assistance. I recently setup SAMBA/Winbind to allow users to login to a Redhat 8 box using their Windows NT Domain credentials. All is working well in that regard. The issue I am having is getting regular UNIX based users to be able to login. The following is my PAM configuration. For example, if I try to login as root, it does not work.

I've installed SAMBA, Winbind etc and everything is working great for users to login with GDM using DOMAIN+username Although this is working, now I can no longer login as a generic Linux user (ex. root). The following is my GDM file from /etc/pam.d/gdm I wonder if someone might have a suggestion as to what it's missing to allow Linux users to login? #%PAM-1.0 auth required

Hi all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. David Morrow

Does anyone have any experience with an Asterisk compatible softphone application which meets the following criteria: 1) Is able to use touch screen rather than mouse for on-screen functions. 2) Has an API which can be used to export Caller ID info to another App on the same compuer. Thanks Bill

Hi. I'm trying to get plugger to work with ogg123 and it does - sort of. No matter what the song is, it plays 24 seconds of the song and then dies. Same spot every time - I've tried with a short song (Janis Joplin Mercedes Benz) and a long song (Cracker Eurotrash Girl). I have this as the mime identification in the plugger rc file: audio/ogg: ogg: Vorbis Ogg audio application/x-ogg: ogg:

Hi all, Has anyone had experience with installing AMP on a soekris box running Astlinux? Is it possible ? Cheers, Callum

Hi all, After pinging the AMP userlist at SourceForge, I got a great step by step explanation as to how to set up AMP for Apache2 (some maybe obvious stuff that wasn't in the Newbie Guide). Thanks to Jason Becker of Coalescent Systems. If anyone needs me to post Jason's instructions here, I can, but they are in a thread called "AMP noob issues with Apache2/Suse9.2" at

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

Hi, I'm using the new AMP which provides a call recording. The options of recording call Always and Never are well working. But how to use the On-Demand option ? Should I press a pad ? Is this configured in the featuremap of features.conf ? Why my modifications in that features.conf have no effects ? Please advice me. Alexis.

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension

After reading about Intel's 'Shevlin Park' project to implement C++AMP in llvm/clang, and failing to find any code for it, I decided to try to implement something similar. I did it as an excuse to explore and hack on llvm/clang, which I hadn't done before, but it's now at the point where it will run the simplest matrix multiplication sample from MSDN, so I thought I might

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I'm in charge of two Asterisk servers. One of these servers is running an Asterisk@Home version under CentOS, the other one is running Debian Sarge. The Debian server has been installed by me using this howto: http://www.astpp.org/index.php?n=ASTPP.DebianSarge-AMP On both servers, the AMP version is 1.10.010. However, the