similar to: REPOST: features.conf *1 Call Recording

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

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 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 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, 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 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 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

I have SAMBA & Winbind working quite well. Users are able to login with DOMAIN+username. With SAMBA sharing using the following config snipit, each user can see their home directory from their Windows machine, but when they try to access the directory, they are prompted for username/password. No combination of username or password seems to work. comment = Home Directories browseable

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

I have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=>#9 in my features.conf, I have Dial(....,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is

hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

Actually the docs for the Queue application say: 'w' -- allow the called user to write the conversation to disk via Monitor 'W' -- allow the calling user to write the conversation to disk via Monitor couldn't get these to work tho. Does this mean I can do one touch recording with agents, or does it mean I can use the monitor() command? Very confusing... Doug.

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 have been using Asterisk for sometime now and have recently come across AMP for the first time. I am wondering if someone could enlighten me a little as to the advantages and disadvantages to using AMP as opposed to the "do-it-yourself" Asterisk? Is this documented someplace? Any advise would be greatly appreciated. David A. Morrow Technical Systems Lead Autodata Solutions

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:

Hi, 1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is

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

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will

Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough. I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover cable). This PBX used to be able to