Hi:
How can I configure sendmail file to asterisk send voicemails to my
mail.sendmail file in /usr/sbin is a read only file.
I'd appreciate any help.
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AstriCon 2008 -
Hi All;
I see that Asterisk has call center, but is it possible to have CRM Integration?
If yes, then how the integration will be? Is it via CTI? From where I can get
the CTI API's to include it in the CRM application and let it communicate with
the CTI server to complete the Integration.
Any
your mail is not clear at all.
if you want to change the path of sendmail ,do this with mailcmd, in the
voicemail.conf,
if you want to send a voicemail to a class of emails, using dbase is more
easier.
let me to know more, about your problem.
--- On Sun, 6/29/08, fateme fatah [EMAIL
Dear Asterisk users,
Voiceroute will be at exhibiting and presenting at the below open source
communications related conferences, Druid meetups speaking about Druid
Asterisk. We would like to meet with other fellow Asterisk enthusiasts who
may be at OSCON LinuxWorld.
Mark Spencer will be
exten = _078.,3,Playback(platna|noanswer)
Thank you everything work perfect now :)
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Register Now: http://www.astricon.net
Hi,
Am Mittwoch, den 25.06.2008, 08:42 -0400 schrieb Jerry Geis:
I am running asterisk from svn check out from yesterday Jun 24.
I started with 1.4.20, then 1.4.21 then svn.
I am getting:
pcm_local.h:389 snd_pcm_channel_area_addr assertion bitsofs %8 = 0 failed
segment fault.
I am
Hello
I'm running Asterisk 1.4.20.1 on a FreeBSD 6.3 host, and unless I'm
mistaken, it seems like /usr/local/etc/rc.d/asterisk script doesn't
make use of /usr/local/sbin/safe_asterisk to restart Asterisk in case
it crashes.
Is this correct, and if yes, why not use it?
Thank you.
Hi,
I've been trying to setup hinting recently on 1.4.20.1, and was
wondering if there is a more elegant way to do the following
piece of dialplan without repeating the hints for every existing
extension/user?
context Main {
hint(SIP/10301) 10301 = call(${EXTEN});
Vincent wrote:
Hello
I'm running Asterisk 1.4.20.1 on a FreeBSD 6.3 host, and unless I'm
mistaken, it seems like /usr/local/etc/rc.d/asterisk script doesn't
make use of /usr/local/sbin/safe_asterisk to restart Asterisk in case
it crashes.
Is this correct, and if yes, why not use it?
Hello all!
I have HFC ISDN Card running in NT mode connected to another PBX what I want
to do is to indicate new incomming call when both B channels are congested
on D channel with CallerID. The other PBX system used to do it with my phone
provider, so that is not a problem. I am using bristuff
Sherwood McGowan wrote:
Gentlemen,
I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime.
This system works fine with 1.2.28, and everything loads fine with no
errors, but when I log an agent in I see the extra message (not in
use) by their listing and they are not rang by
On Sun, Jun 29, 2008 at 04:32:40PM +0200, michael_t Gazeta.pl wrote:
Hello all!
I have HFC ISDN Card running in NT mode connected to another PBX what I want
to do is to indicate new incomming call when both B channels are congested
on D channel with CallerID. The other PBX system used to do
Debug show completely nothing. I think there is something missing in my
config files. Should it work just with callwaiting and callwaitingcallerid
in zapata.conf or something more? my zapata.conf looks like:
switchtype = euroisdn
signalling = bri_net
pridialplan = dynamic
prilocaldialplan = local
We'd be happy to put our appliance up against the rest - I've cc'd in
the asterisk list as it doesn't have to be a commercial magazine or
something like that but if anyone wants to organize an 'asterisk
appliance shootout' just sit down and post a methodology or what exactly
the want to test for
On Sun, Jun 29, 2008 at 7:02 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Sherwood McGowan wrote:
Gentlemen,
I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime.
This system works fine with 1.2.28, and everything loads fine with no
errors, but when I log an agent in I see
Hi All;
How to increase the waiting time between entering the digits for the analoge
phone device that is connected to fxs?
Is it by DigitTimeout? But how it will be apply for analoge station if the user
just pickup the handset and dialed the number?
Any help?
Regards
Bilal
On Jun 29, 2008, at 6:35 PM, bilal ghayyad wrote:
Hi All;
How to increase the waiting time between entering the digits for the
analoge phone device that is connected to fxs?
Is it by DigitTimeout? But how it will be apply for analoge station
if the user just pickup the handset and
I've got a unique situation and think it may be the lack of the Hangup command
in the dialplan that is creating the issue.Can anyone elaborate on why it
is, or is not, important to use hangup in the dialplan. Presently I don't have
the first instance of it in my dialplan, however, I see
In a 1.2 release of asterisk, I've had no problem connecting to a Broadvoice
SIP peer, to allow routing outgoing calls from Asterisk to Broadvoice. Now,
with the same SIP configuration, I cannot establish the peer. I've enclosed a
SIP log in the hope that someone can help me analyze this
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