similar to: Dial Command Reference for SIP channel

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 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 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 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 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

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, 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

Hi Akk, If this has been discussed/done then apologies be-4-hand. I did not find it in the Wiki or the Archives. Here's the question. We have incoming PRI lines, all on the same main number. An attendant is supposed to handle all incoming calls. Now, let's say I have a multi-line SIP phone. For argument's sake (and to keep it simple) say I only have two lines. We'll call them

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

Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out. [outbound-local] exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9ZXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten =>

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call

Friends I am looking at Rcpp and I am a bit stuck on a simple matter. (I am calling R from c++, if there is a better way...) Given this simple example using the TTR package and the SMA function which returns a simple moving average.... Rcpp::NumericVector rv; for(int i = 0; i < 100; i++){ rv.push_back(rand()); } Rcpp::Environment TTR("package:TTR");