Displaying 20 results from an estimated 3000 matches similar to: "features.conf *1 Call Recording"
2006 May 10
2
REPOST: features.conf *1 Call Recording
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 =>
2006 Jan 30
2
Most Popular FREE SoftPhone for Windows
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
2005 Jan 11
3
AMP Anyone?
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
2006 May 12
3
Dial Command Reference for SIP channel
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
2005 May 18
7
Soft Phone
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
2005 Feb 24
1
Winbind Authentication on Redhat & Home Directories
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
2003 Jul 18
16
Call Transfer
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
2002 Nov 18
2
Problem with SAMBA & Winbind
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
2002 Nov 15
1
Winbind and Samba
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.
2002 Nov 18
1
Help with PAM Config
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
2007 Aug 03
1
Password Encryption
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
2006 Mar 07
2
pap2 Dial plan
Hi
i am using pap2 phone adaptors as clients to connect to asterisk server
i am able to make calls but i cannot access voice mail using phone
or start recording while call is in progress
and when i place a call to local sip extension there is a long pause ( 15
sec )
before the call gets dialled
i assume that the problem would be due to the dial plan in PAP2
if so please help me changing it
2006 Apr 02
2
DID registration status
HI
I have two sip accounts from two different ITSP's both configured on
asterisk server. how can i know if these accounts have been successfully
registered ?
i generally look at the /var/log/asterisk/full
suggest me if there are better way of doing this
thanks
Giridhar Bandi
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2006 Mar 30
9
How is Teliax ?
Hi
I am looking at purchasing some DID lines from Teliax to install it on my
asterisk.
i would like to know some feed back on "Teliax" before i purchase.
suggest me if there are better sevice providers.
thanks
Giridhar Bandi
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2006 Mar 08
3
RES: pap2 Dial plan
Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/x-pkcs7-signature
Size: 3050 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/3396d198/smime.bin
2006 Apr 18
3
IVR: playing multiple streams simultaneously?
Hi all,
I'm setting up an IVR using Asterisk.
Is there a way to have two streams played to the caller at the same
time: for instance, one constant flow of background music, and the IVR
contents at the same time? I've looked for solutions using (E)AGI and
other things but nothing seems to work. Googling around and reading the
list has not been helpful either...
Thanks for your help,
2008 Mar 13
5
Newbie One-touch Recording: Does not work
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:
2005 Jan 25
2
New native assisted transfer (atxfer) usage info required
Hi, I would like to use the new atxfer (native assisted transfer, see
mantis item #3241) , but I've partially been able to
make it work.
I can receive a call and then having the caller hear MOH while talking
with another extension (the one I want to transfer to), but then I can't
make the caller and the trasferred talk hanging up or pressing any key
combination I'm aware of.
My
2005 Jan 28
1
Minimum Setup
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
2005 Jun 29
1
AMP or Asterisk
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