Hi folks!
I want to get the talk time after a call, could you tell me some ways how to
do that?
Thanks !
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE
On 02/09/2010 05:01 AM, lemonash wrote:
I want to get the talk time after a call, could you tell me some ways
how to do that?
What does that mean?
--
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1 678-954-0670
Direct : +1 678-954-0671
Web: http://www.evaristesys.com/
--
On Tue, Feb 9, 2010 at 5:01 AM, lemonash lemonash2...@gmail.com wrote:
Hi folks!
I want to get the talk time after a call, could you tell me some ways how to
do that?
Thanks !
Check your CDRs or some phones keep record of how long each call
lasted. You could go low tech and keep a stop
Set up the CDR.
http://www.voip-info.org/wiki/view/Asterisk+cdr+csv
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: 09 February 2010 10:10
To: Asterisk Users Mailing List - Non-Commercial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Hi
I have searched a bit for information regarding the status on the
dialplan in lua (pbx_lua.so). I know that 'hint' won't work and has to
be put in the regular extensions.conf/ael. Are there any other
considerations? Features that are not
Am 08.02.2010 21:15, schrieb Philippe Sultan:
Philippe, what exactly is a playback channel? Is it a pseudo participant
playing back the announcements?
Yes. Announcements are played to the conference members by creating a
channel of type 'Bridge' which streams the sound files.
thanks
Thanks Josiah Bryan,
I do not have any dns server running on my asterisk server, we do have an
external DNS server working in the data center; the IP of this dns server is
10.4.1.5...
Following you will see my main configuration:
/etc/resolv.conf:
domain localdomain
search localdomain
Hello,
I have been trying to setup asterisk (1.6.2.0) to receive fax. I am
able to receive faxes sent from a zoiper softphone connected to
asterisk.
I have some DID numbers (with T.38 support) forwarded to my asterisk
pbx. I am not able to receive faxes from these numbers. The error I
get on the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
I have been trying to setup asterisk (1.6.2.0) to receive fax. I am
able to receive faxes sent from a zoiper softphone connected to
asterisk.
I have some DID numbers (with T.38 support) forwarded to my asterisk
pbx. I am not able to receive
Not sure it is relevant, however, I have an E52 I use occasionally with
my * and I've found that without an active SIM in the phone the SIP
profile will ring silently.
I'm sure there's a way to fix it I just haven't been bothered enough to
work on it.
sl
From:
On Tue, Feb 09, 2010 at 07:42:48AM +0300, Muro, Sam wrote:
Hi Team
Can someone advice me on how i can lower the load average on my asterisk
server?
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.10.1
asterisk-1.4.25.1
2 X TE412P Digium cards on ISDN PRI
Im using the system as
On Tue, Feb 9, 2010 at 6:24 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
I wonder what mute should mean. Does it mean that the participant will
not receive any media, or that media sent by the participant will be
ignored, or both?
Please post your discoveries to:
Am 09.02.2010 15:35, schrieb David Backeberg:
On Tue, Feb 9, 2010 at 6:24 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
I wonder what mute should mean. Does it mean that the participant will
not receive any media, or that media sent by the participant will be
ignored, or both?
On 9 February 2010 06:42, Muro, Sam resea...@businesstz.com wrote:
Hi Team
Can someone advice me on how i can lower the load average on my asterisk
server?
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.10.1
asterisk-1.4.25.1
2 X TE412P Digium cards on ISDN PRI
Im using the
On Tue, Feb 9, 2010 at 10:26 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
Answering myself: muting means that the participants voice is ignored.
Thank you for updating the wiki and the list.
I looked into this when I was having problems with early 1.6.0.*
MeetMe(), specifically the
Come on, was that necessary? He was asking for help and considers it an
important issue... If you want to chastise the guy at least offer up a
solution for him...
Sam - I have never tried this solution but Sangoma has a reference to
this.
Overlap receiving timeout, plus dialplan latency, causes network to retry
SETUP
https://issues.asterisk.org/view.php?id=16789
This patch removes the requirement that some may have found that you need to
insert a Proceeding() statement very early in your dialplan, otherwise an
inbound overlap
If it works, I wouldn't care it is was in Chinese; That's what google
translator is for...
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon
Sent: Tuesday, February 09, 2010 11:24 AM
To:
test--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
fail.
On Tue, 9 Feb 2010, aster...@opensourcesolution.in wrote:
test
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Thanks. It's working now.
In my sip.conf I had 't8pt_udptl=yes'. I changed it to
't8pt_udptl=yes,redundancy,maxdatagram=400' and it started working.
On Tue, Feb 9, 2010 at 6:22 PM, Tommy Botten Jensen
tommy.jen...@freecode.no wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
I have
Scenario:
[Asterisk Server] on routed/public IP
\/ /\ \/ /\ \/ /\ \/ /\ \/
[Draytek Router] -- internal IP's
\/ /\ \/ /\ \/ /\ \/ /\ \/
[internal Network 192.x.x.x]
[IP10s] + [IP10s] + [Softphones]
Everything is good as long as only *1* phone is registered from the
internal
hi,
i havent spent that much time with asterisk lately, but still wanted
to gather information on how to initiate a call:
1) fact
what i know which is possible:
- via call-file
- via (sip)-client
- AGI
2) desired
- URL
-- is this possbile?
3) others
-- whats missing here?
thx
--
#2 is possible through AMI/ajax interface. See voip-info.org.
#3 is pretty much covered by AMI. Anyway you can think of not mentioned in
#1 or #2 would (IMO) probably include AMI usage.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
thx danny, i checked ur link, but not sure for what to search in terms
off http/js/ajax...
my application is a rails, so adhearison / rami apps sound goo dto me,
but if possible i would go without the roundtrip to my application
server
tom
On Tue, Feb 9, 2010 at 1:59 PM, Danny Nicholas
On Tue, Feb 09, 2010 at 01:48:35PM -0500, tom wrote:
hi,
i havent spent that much time with asterisk lately, but still wanted
to gather information on how to initiate a call:
1) fact
what i know which is possible:
- via call-file
- via (sip)-client
- AGI
Inside Asterisk? Surely you
hi, im trying to figure out whats the easiest way to achieve the
following scenario:
- inhouse asterisk
- inhouse dynamic web/html-app
users of this application have a sip-account. the app is similiar to a
CRM system. so i want to put up a link , and if they click the link it
should ring their
Hello list,
I've got a client who's weak sip passwords are being guessed by remote
entities who then connect to their server and use it to wardial large
swaths of numbers. When they start receiving complaints, they call me
and I add the ip address of the remote user to the iptables drop
Here it is in a nutshell (in Perl, you can alter for your preferred
language)
my %respa = $astman-sendcommand( Action = 'Originate',
Channel = SIP/$fromval,
Exten = $toval,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
As Tzafrir mentions, the AMI can be used here. It also has a
web-interface, so you can do it via a series of URLs. Look up AJAM which
might be what you are looking for.
- - Tommy Botten Jensen
tom skrev:
hi, im trying to figure out whats the
On Tue, 9 Feb 2010, Warren Selby wrote:
Is there some logging capability that allows me to see every IP address
of every sip registration attempt, along with details about the sip reg
attempt (I.e user name tried, success or failure, user agent, etc). I
haven't found a way to do this yet,
YC Nyon wrote:
hi,
I'm been successful in making calls to another local extension using
Nokia E71. However calling the E71 from another ext. (X-lite) is not
successful. There is a ringing tone from the caller side but the E71 is
silent.
Tried disabling the NAT (dunno whether that
I have a strange callerid problem. All my SIP phones display the
correct name of the caller but the number is always the number of the
extension that was called. If I do a NoOp on the dialplan I can see
that both name and number are correct.
The call log in my phone records all
Warren Selby wrote:
Hello list,
I've got a client who's weak sip passwords are being guessed by remote
entities who then connect to their server and use it to wardial large
swaths of numbers. When they start receiving complaints, they call me
and I add the ip address of the remote
Hi all,
I am configuring asterisk as a prepaid calling card. I am getting different
local rate from my ISDN provider e.g 0.002 for landline and 0.13 for mobile
etc. In this case I thing I have to say my asterisk/a2billing to bill based on
local access number. so How can I retrieve called
On Tue, Feb 09, 2010 at 03:40:48PM -0600, Danny Nicholas wrote:
Here it is in a nutshell (in Perl, you can alter for your preferred
language)
my %respa = $astman-sendcommand( Action = 'Originate',
Channel = SIP/$fromval,
Figuring out
Hi Steve
Even though you shouldn't have to, have your rebooted? 200 days of
uptime and this just started?
It seems this problem is common as i have three boxes of the same capacity
with exactly the same problem. So reboot should only solve the problem for
a while
Have you recently updated
hi All,
its probably very simple but i can't find the way to it.
i have some b410p cards and use them to connect to ISDN2, this works OK
for calling but i need to have 1 line to be send to the fax machine.
the fax machine is a modem connected on another machine with hylafax.
as far as i can
Hi Team
Can someone advice me on how i can lower the load average on my asterisk
server?
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.10.1
asterisk-1.4.25.1
2 X TE412P Digium cards on ISDN PRI
Im using the system as an IVR without any transcoding or bridging
Hi,
Barring WW (wifi woes), I will be broadcasting live from the HD
Communications Summit this Friday. Usually we begin at 12 Noon EST but
we may start earlier so please check the site, IRC, Twitter or
Facebook for the exact start time. If any of you are planning to be
there, please email me if
Hi Team
Can someone advice me on how i can lower the load average on my
asterisk server?
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.10.1
asterisk-1.4.25.1
2 X TE412P Digium cards on ISDN PRI
Im using the system as an IVR without any transcoding or bridging
41 matches
Mail list logo
