Hi,
When I need to route calls depending on the number of (incoming and
outgoing) calls a SIP device is currently handling, I mostly use
function SIPPEER and its curcalls option.
I can read and there references to function GROUP for the same usage,
but I intuitively thought that though this
Hi List,
Please tell me which ports should be required open for communication with
asterisk. like 5060 for sip calls, 4569 for IAX, 10,000 to 20,000..
Apart from these ports what else is required ?
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Hi,
That depends on what else your asterisk is doing i.e if an AMI-based code
is running then AMI port needs to be open as well. It also depends what
other appliactions are running on asterisk-box which require port opening
i.e apache or mysql etc.
Regards,
Sammy
On Mon, Dec 12, 2011 at 3:21 PM,
Nothing?
On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote:
Hi There,
I'm still having this problem, Does somebody know what can be
happening?
Regards.
On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote:
Hello,
The exten is the parameter passed to
Hi Albert,
we currently use QueueMetrics to monitor and report on call center
statistics...
regards
Anthony
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New to Asterisk? Join us for a live
Hi Sammy,
Thanks for fastest reply. I to know just for calling time which port's
should asterisk need to be open only
On Mon, Dec 12, 2011 at 4:03 PM, Sammy Govind govoi...@gmail.com wrote:
Hi,
That depends on what else your asterisk is doing i.e if an AMI-based code
is running then AMI
Danny,
Why would you think this is a circumvent? I'm using a nice feature of 1.8
where I can create any CDR field I like and populate it by using the
CDR(fieldname) function. While all other fields that I created are populated
properly (however before the 'dial' commences) it seems like at
Hi guys,
I have following problem. For statistical reasons I need to know what
was initiall number dialled by customer. I have 2 premium numbers, for
which customers are billed differently per minute. But in my CDR table i
can see only last dialled extension from voice menu. In this example
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 12/11/2011 10:59 PM, Mike Diehl wrote:
Should I go to 1.8.x? Or all the way up to 10.x? This is a
production system and I can't afford to be testing code.
The 1.8 series is the current LTS release.
Barry
-BEGIN PGP SIGNATURE-
hello ,
I have been working hard to solve the issue of custom CDR in the Asterik
with Mysql but in vain.
I searched google for complete 2 hours but in vain.
What i want to achieve is CDR(customcolumn)=anyvaluealthough we can
achieve it through other ways like making a script that runs when
2011/12/12, Mike Diehl mdi...@diehlnet.com:
Hi all,
I have 2 servers running 1.6.2.9 and I'm about to build a third server.
This
suggests the possibility of doing a rolling upgrade of all of my servers.
This brings up the question of what version to install and upgrade to. I
don't have
Are you using FreePBX or another packaged Asterisk?
Sent from my iPhone 4S
On Dec 12, 2011, at 9:23 AM, silent sayz silent.s...@gmail.com wrote:
hello ,
I have been working hard to solve the issue of custom CDR in the Asterik with
Mysql but in vain.
I searched google for complete 2
Hi All,
I have installed centos 5.6 32 bit on xeon server and i have also installed
latest version of asterisk 1.6 and dahdi as well.
I want to install chan_ss7 for this server and I want to know about the
following device.
Digium TE420B
I dont know much about the configuration files for Digium
Hi!
I am using Asterisk 1.6.2.20 with elastix
-- Forwarded message --
From: Robert-IPhone rhuddles...@gmail.com
Date: Mon, Dec 12, 2011 at 5:45 PM
Subject: Re: [asterisk-users] MySql Custom CDR issues
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
I installed asterisk addons package and it is solved. Thank you.
On Mon, Dec 12, 2011 at 5:55 PM, silent sayz silent.s...@gmail.com wrote:
Hi!
I am using Asterisk 1.6.2.20 with elastix
-- Forwarded message --
From: Robert-IPhone rhuddles...@gmail.com
Date: Mon,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, December 12, 2011 8:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What version to upgrade
On Mon, Dec 12, 2011 at 12:26 PM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, December 12, 2011 8:27 AM
To: Asterisk Users Mailing List -
Asterisk uses libcap to do root-like things when running as non-root.
Setting the DSCP/QoS value of packets requires root access, but Asterisk seems
to manage just fine using libcap (not libpcap, that is different).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Leads to the next question - has anybody tested SNMP using non-root Asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, December 12, 2011 9:41 AM
To: Asterisk Users Mailing
On 12/12/2011 09:26 AM, Danny Nicholas wrote:
I'm wondering if the bind 161 as root statement is a mis-statement or
if not, maybe somebody like Tzafir can explain why since none of the
other Asterisk binds require root access (this message is still in
10.0-rc3).
This is accurate. Non-root
Hello all,
I have recently upgraded to version 1.8.7.2 and have started to see the
following errors in the logs:
[ Dec 12 16:10:38] ERROR[10223] app_voicemail.c: IMAP Error: SELECT failed
[ Dec 12 16:10:38] ERROR[10223] app_voicemail.c: IMAP Error: must be in
SELECTED state
They are not
On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote:
Hello all,
I have recently upgraded to version 1.8.7.2 and have started to see the
following errors in the logs:
From what version?
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out
Please start a new thread for new conversations.
On Monday 12 Dec 2011, Albert wrote:
I have following problem. For statistical reasons I need to know what
was initiall number dialled by customer. I have 2 premium numbers,
for which customers are billed differently per minute. But in my CDR
1.8.7.0 ... am using Zimbra as the backend IMAP storage.
--
Thanks, Phil
- Original Message -
On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote:
Hello all,
I have recently upgraded to version 1.8.7.2 and have started to see
the following errors in the logs:
From what version?
Hello again,
Still with the same issue of dahdi off hock state.
I changed from:
dahdi - 2.2.0.2 to 2.6.0.
astersik - 1.4.26.2 to 1.4.29.
If I restart Asterisk, the problem persists. If I restart dahdi, and
after start asterisk,
the issue disappears for a while. Thus the problem
Hmmm, just tried leaving a voicemail on a new mailbox where the imapfolder
contains a space in the name and it errors; so that could be the cause of it
all. Is is valid to have a space in an IMAP folder name ?
--
Thanks, Phil
- Original Message -
1.8.7.0 ... am using Zimbra as the
Generally speaking, no. if you need the space, use quotes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Sent: Monday, December 12, 2011 11:12 AM
To: Asterisk Users Mailing List -
Okay, though removing the space and reloading the module still throws the same
error messages.
--
Thanks, Phil
- Original Message -
Generally speaking, no. if you need the space, use quotes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Any suggestions from people who have done this before?
Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCSA, Security+, A+
Linux+, Network+, Server+
A.A.S. Information Technology
.
www.impalanetworks.comhttp://www.impalanetworks.com/
P: (505) 327-7300
F: (505) 327-7545
--
Build Asterisk with ODBC support and then use the ODBC functions to do the
database dips.
On Dec 12, 2011, at 13:44, Douglas Mortensen d...@impalanetworks.com wrote:
Any suggestions from people who have done this before?
Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks
Well, I was wrong. The messages went away for a day, then came back. I
am now rebuilding the server using an older motherboard. Hopefully that
will solve the problem.
On 12/9/2011 4:09 PM, Brent Davidson wrote:
For the sake of posterity, I'm posting this solution:
When I checked the
Hi everyone,
We are looking to develop our own call centre application (HTML5,
real-time, shiny GUI, easy access, etc...) on top of Asterisk. We are tired
of using the proprietary packages that currently exist due to no proper
support, expensive licensing costs, ugly GUIs, and closed nature of
Hi List,
Has anyone heard of an ATA device that supports TCP TLS? Not much
comes up in searching, thought to check here for some device
suggestions.
TIA,
Skyler
--
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-- Bandwidth and Colocation Provided by
The OBI110 ($45 USD) supports both of these.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler
Sent: Monday, December 12, 2011 1:55 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ATA with
On Monday 12 December 2011 4:28:17 am Harel Cohen wrote:
Danny,
Why would you think this is a circumvent? I'm using a nice feature of 1.8
where I can create any CDR field I like and populate it by using the
CDR(fieldname) function. While all other fields that I created are
populated
On Monday 12 December 2011 6:39:34 am Barry L. Kline wrote:
On 12/11/2011 10:59 PM, Mike Diehl wrote:
Should I go to 1.8.x? Or all the way up to 10.x? This is a
production system and I can't afford to be testing code.
The 1.8 series is the current LTS release.
Barry
Well, that
Hi All,
There are a lot of existing projects for configuring Asterisk via GUI,
so instead of trudging through them all, I'm hopeing to get some
guidance.
My architecture is ITSP based, we supply hosted PBX's to business
customers. A few systems are dedicated PBX's but the majority are
Hi folks.
I've got a problem dialing with my new Snom M9 via TLS on asterisk 1.8.7.1 .
Registration works like a charm - the phone becomes 'AVAILABLE'.
An INVITE is responded by a 401 to be expected, but then asterisk closes the
TLS connection upon the Snom's ACK.
The authenticated INVITE the
Hiya,
SIP Messaging is implemented in asterisk-10...
The only documentation I can find talks about a patch and is pretty
old:http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging
http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging
Like anything on voip-info.org it's horrible
I think it only works with certain soft phones. I tried Aastra and it
doesn't work. But EyeBeam soft phone receives messages.
-Bruce
On Mon, Dec 12, 2011 at 6:40 PM, Jay R. Worthington
jayrworthing...@gmail.com wrote:
Hiya,
SIP Messaging is implemented in asterisk-10...
The only
hello:
you can refer this link:
http://mirror.su.lt/voip-info/wiki/view/Asterisk+ss7+channels.html
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
Date: Mon, 12 Dec 2011 20:21:36 +0530
From:
Is there a need to do it within the dialplan? If not you will find it easier to
do it within AGI. Either connecting directly to the DB or in our case our
developer build a web service which I make SOAP calls to.
Nick.
From: asterisk-users-boun...@lists.digium.com
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