similar to: AMP - recording call

Hi, I am trying to use the attended transfer. So I put this in my feature.conf: [general] [featuremap] atxfer => *0 blindxfer => #0 I completly restart asterik, and not just make a RELOAD. But during a call, when I press # it runs a blind transfer and if I press * I am disconnected. I am using the CVS version of * get as explain here

Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (green part). I make modprobe zaptel && modprobe wctdm without

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

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:

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

Hi all, If I activate recording for an extension everything is OK. but If I activate call recording on demand i am non able to start recording In principle I should have to press *1, as indictaed in features.conf (I am using almost last asterisk code, updated 2 days ago from svn, version SVN-branch-1.2-r39379M ) Actually it produce no effect at all I am using FreePBX interface, and I saw

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

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I

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

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

MixMonitor seems to work: -- User hit '*3' to record call. filename: auto-1250792853-24-22 == Begin MixMonitor Recording SIP/snom2-084c4ec8 /var/spool/asterisk/monitor/auto-1250792853-24-22.wav exists now. Recording a call without mixing fails. > User hit '*1' to record call. filename: wav,auto-1250793354-24-22,m TOUCH_MONITOR_OUTPUT is set to

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

Hello all, The "Secret Agent" final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script

Ladies and Gentleman, I have setup Asterisk and AMP. They are working with out error. But now I need to get everything going. I have Voicepluse and they give sample iax.conf and extensions.conf files but that does me a little good as I am using AMP. Is there some docs somewhere on how to confgure once I have AMP up and running? I am not a telephony guy and alot of this looks like greek to

I've just started using AMP and found that I have a problem with escaped characters in config files. In particular, I have a custom config item that needs a semicolon in... SetVar(_ALERT_INFO=info=auto-answer;delay=1) To get the part of the line after the ; to be accepted by Asterisk as a non-comment it needs to be escaped with a backslash, but I have found that I need to put two

Hi All, Does anyone know if AMP can work with Fedora Core 3? I've tried to install AMP, and * on FC3, but found FOP failed to start when you use the command amportal start. Please advice if anyone got the solution. Chichi --------------------------------- Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more. -------------- next part -------------- An

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, I can't figure out how to change User Login and Password for AMP. By default it is user:admin and password:maint. Anybody knows how to do it? Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/c8ef57b2/attachment.htm