On Wed, May 10, 2017 at 10:11 AM, Steve Edwards
wrote:
> I have a 'time and attendance' application. Think janitorial or security
> kind of thing where an employee goes from location to location.
>
> They're supposed to 'clock in' when they get to a site using a phone at
> that site to prove they
On Thu, Apr 9, 2015 at 12:23 PM, Derek Andrew wrote:
> SNOM phones can be configured using files on a TFTP server.
>
> On Thu, Apr 9, 2015 at 11:14 AM, jg wrote:
>
>>
>> Does anyone know how to program Snom phones using a Mac addresses like
>> what is done with the Ciscos. I have about 50 exte
On Thu, Jun 27, 2013 at 10:34 AM, A J Stiles
wrote:
> On Thursday 27 June 2013, Eric Cooper wrote:
>> I'd like to protect my expensive Digium FXO cards from spikes on my
>> three incoming PSTN lines. Does anyone have any recommendations?
>
> Does your telco not fit surge suppressors to the NTE as
t; At 9.4 cps 96963 calls were executed with 0 failed calls. Where is the
> concurrent call figure in this tool? Please forgive me still getting
> use to it :).
>
> In regards to hardware transcoding cards for SIP protocol. Please let
> us know of some digium solutions. Again, we woul
On Sat, Mar 23, 2013 at 12:06 PM, Joshua Colp wrote:
> Nick Khamis wrote:
>>
>> Oh no secret. Some things I do is increase the ulimit size. I was
>> wondering if there was a way to increase allocated memory. I have been
>> reading about a -p option but when I start asterisk using "asterisk -p
>> -
On Sun, Mar 10, 2013 at 11:37 AM, Paulo Victor Fernandes da Silva
wrote:
> hello guys,
>
> I'm working on a federal university at Brasil, we already have an openLdap
> with all users and this base is used to authenticate several services like
> email, vpn, wireless (RADIUS), and we have also Shi
On Thu, Jan 17, 2013 at 6:32 PM, Bryant Zimmerman wrote:
> Hey all
>
> For some reason the mailing list is sending all messages from the sending
> party.
> This makes it less than ideal when responding; as selecting reply goes to
> the person and not the list.
> Can we have it set back to the old
On Thu, Jan 17, 2013 at 3:05 PM, Joe Ruffolo wrote:
>
> Hi all! In need of some serious help. We currently run Trixbox and Cent Os
> on a 2u server for our small business’s phones system.
>
>
>
> We are using some Polycom Soundpoint IP phones. The whole thing came
> crashing down over the Holidays
On Thu, Jan 17, 2013 at 3:02 PM, Bryant Zimmerman wrote:
> Hey all.
>
> RE: Conf Bridge.
>
> I am looking into a project that would need 8 to 10 thousand parties in a
> single conference.
> Most would be on mute but 5 to 6 would be presenters.
>
> Is the new conf bridge solid enough to handle this
On Sat, Nov 3, 2012 at 2:16 PM, Danny Dias wrote:
> Hello,
>
> I wonder if Digium provides support for Asterisk OpenSource versions as an
> anual fee or something?
>
> For example, if i download Asterisk 1.8.X (Certified or not...) can i buy
> support from Digium to maintain and help on possible f
On Fri, Oct 19, 2012 at 11:28 AM, Eric Wieling wrote:
> I'm setting up a test server with a Digium TE122 and am getting the following
> error on the console, spewing as fast as it can. Does anyone have any idea
> what this error might be?
>
> [Oct 19 11:24:53] NOTICE[2076]: chan_dahdi.c:3108 my
On Mon, Oct 15, 2012 at 3:07 AM, sudeep melekar
wrote:
> hello,
> i want to install asterisk 1.8 in a single directory "myasterisksetup"
> i.e after asterisk installs it put some of it's installation files in
> different directories
> e.g /var/log/asterisk
> /var/run/asterisk
> and many more
>
> i
On Tue, Oct 2, 2012 at 8:04 PM, Steve Edwards wrote:
> On Tue, 2 Oct 2012, Mitch Claborn wrote:
>
>> I'd like to be able to use the same config files in CVS and have the
>> differences resolved at run time, based on host name of the asterisk server.
>>
>> Another idea would be to write a simple pe
On Fri, Sep 28, 2012 at 5:27 PM, Adam Moffett wrote:
> I was asked today if we could somehow have a trainee on the phone with a
> supervisor conferenced in, but somehow have it so anything the supervisor
> says is only heard by the trainee and not the customer.
>
> Is there a feature like that?
>
On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer wrote:
> 2012-08-28 16:44, Andrew Latham skrev:
>>
>> On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer wrote:
>>>
>>> Hi,
>>>
>>> I've used the shells-script at the end of this email to generate
On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer wrote:
> Hi,
>
> I've used the shells-script at the end of this email to generate 8khz mono
> wave-files for asterisk from a 144 khz recording.
>
> The script does two things: resample & normalize the audio volume.
>
> Anyone like to share their recom
On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro wrote:
> Hoo-hah. It registers. Progress!
>
> Now... media. Or not.
>
> On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp wrote:
>> - Original Message -
>>> >
>>> > The complete URL to use is http://>> > host>:8088/ws
>>> >
>>> > Note the /ws at th
On Mon, Aug 20, 2012 at 10:52 AM, Joshua Colp wrote:
> - Original Message -
>> Joshua
>>
>> Can you copy and past into a wiki page for everyone's benefit? Maybe
>> https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or
>> like page would be good.
>
> If this thread has taught
On Fri, Aug 17, 2012 at 4:07 PM, Andrew Latham wrote:
> On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp wrote:
>> - Original Message -
>>> On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro
>>> wrote:
>>> > I see no indication of how to do this in sip.co
On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp wrote:
> - Original Message -
>> On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro
>> wrote:
>> > I see no indication of how to do this in sip.conf, and when I start
>> > Asterisk, it doesn't wait on port 80.
>> >
>>
>> Websocket support is being ac
On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro wrote:
> I see no indication of how to do this in sip.conf, and when I start
> Asterisk, it doesn't wait on port 80.
>
> Greetings,
>
> --
> Juan Carlos Castro y Castro
> Instant Solutions - Telefonia Gerando Resultado
> http://www.instant.com.br
> Prin
t;
>> best regards,
>> Sven2012/8/13 James Mortensen a-cti.com>
>> Andrew Latham gmail.com> writes:
>> >
>> > On Mon, Aug 13, 2012 at 2:58 PM, James Mortensen
>>
>> > a-cti.com> wrote:
>> > > Hello,
>> > >
>>
On Mon, Aug 13, 2012 at 2:58 PM, James Mortensen
wrote:
> Hello,
>
> I'm trying to register a user using sipml5 on Asterisk 11. I followed the
> instructions here:
> http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets
>
> I added transport=ws to my sip.conf file:
>
> [3002]
> use
On Sun, May 6, 2012 at 12:42 PM, Greg Woods wrote:
> I have a Digium TDM400P card that appears to have died. The first noted
> symptoms were that dahdi would fail to reload on boot. On closer
> inspection, the card looks totally dead; no lights on at all. I have
> tried moving it to a different PC
On Wed, Mar 21, 2012 at 3:04 PM, Jonas Kellens wrote:
> Hello,
>
> when generating backtrace I get following output :
>
> [root@sip ~]# gdb -se "asterisk" -ex "bt full" -ex "thread apply all bt"
> --batch -c core.sip-2012-03-21T10\:57\:29+0100 > /root/backtrace.txt
> asterisk: No such file or dire
On Wed, Mar 21, 2012 at 3:10 PM, Paolo Supino wrote:
> H Andrew
>
> Your solution is the simplest I received and so I tried implementing
> it only to discover that it doesn't work as expected...
>
>
>
>
>
>
> TIA
> Paolo
Check your Dial() options... "Verify your options to you dial syntax
and a
On Wed, Mar 21, 2012 at 8:45 AM, Tony Mountifield wrote:
> Over the years I have experienced a few interrupt issues when using some
> of the Digium E1/T1 cards with Zaptel drivers, and usually resolved them
> by disabling USB devices in the motherboard BIOS settings.
>
> Now more and more systems
On Wed, Mar 21, 2012 at 8:27 AM, Paolo Supino wrote:
> Hi
>
> I was asked by our development departement to setup asterisk in a
> manner that if someone calls an extension in the department that was
> was only configured, but a handset was never attached to it to fall
> back to a default extensio
On Mon, Feb 6, 2012 at 8:24 PM, Josh wrote:
> In short - is this module essential for the running of Asterisk? What is its
> function? Is there a help/list where I could find a description of what it
> does? Thanks!
The primary goal was to upload audio for IVRs in the Asterisk GUI.
--
~ Andrew
On Thu, Dec 22, 2011 at 3:42 PM, Kevin P. Fleming wrote:
> On 12/22/2011 12:02 PM, Shaun Ruffell wrote:
>>
>> On Thu, Dec 22, 2011 at 02:54:05PM -0300, Andrew Latham wrote:
>>>
>>> On Thu, Dec 22, 2011 at 2:48 PM, Shaun Ruffell
>>> wrote:
>>>>
On Thu, Dec 22, 2011 at 2:48 PM, Shaun Ruffell wrote:
> On Thu, Dec 22, 2011 at 06:33:44PM +0100, Olivier wrote:
>> Hi,
>>
>> Testing 1.8.8.0 (with Dahdi 2.5.0.2 and asterisk-gui 2.1.0-rc1), I'm
>> seeing this on my console:
>>
>> WARNING[25363]: config.c:1208 process_text_line: Unknown directive
On Thu, Dec 22, 2011 at 2:33 PM, Olivier wrote:
> Hi,
>
> Testing 1.8.8.0 (with Dahdi 2.5.0.2 and asterisk-gui 2.1.0-rc1), I'm
> seeing this on my console:
>
> WARNING[25363]: config.c:1208 process_text_line: Unknown directive '#'
> at line 1 of /etc/asterisk/../dahdi/system.conf
>
> This warning
On Wed, Dec 21, 2011 at 12:03 PM, Bryant Zimmerman wrote:
> We have written some monitoring and stat collection scripts that use
> asterisk -rx "command" The script runs once a min and logs data and posts
> any critical notifications. Everything is working well with this method but
> we get the
On Thu, Dec 15, 2011 at 3:39 PM, Carlos Alvarez wrote:
>
> On Thu, Dec 15, 2011 at 11:33 AM, Tarek Sawah
> wrote:
>>
>>
>> Hello List,
>> I have customer with a 40 Agents call center. and is looking to install a
>> PBX switch that can serve those agents.
>> As per my experience i suggested Asteri
On Thu, Dec 15, 2011 at 11:05 AM, Vieri wrote:
> Hi,
>
> I have a new Digium TE205P 2-span E1 card I just installed on a server.
>
> As soon as I boot the machine, the card's leds flash red (ports 1 and 2) -
> even when in the BIOS.
>
> That's not good, right?
>
> I don't have another machine to
On Mon, Dec 12, 2011 at 12:26 PM, Danny Nicholas 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 - Non-Commercial
On Wed, Nov 30, 2011 at 6:20 PM, Ferdinand Babas wrote:
> Hi All,
>
> I've been trying to find a solution that would allow our sip phones to
> communication with walkie talkies. Our setup is that we have sip phones
> setup in 2 locations, headquarters and dome. We can communication from
> headqu
On Wed, Nov 16, 2011 at 10:46 AM, Hans Goossen wrote:
> Hello group,
>
> I have this situation:
>
> I have several contexts with a few extensions each one. I need to give every
> context a limited quantity of minutes they can use. All the extensions in the
> context will share the same "bag" of
On Sat, Oct 29, 2011 at 3:14 PM, Eric van der Vlist wrote:
> Hi,
>
> Xorcom astribanks get initialized straight on when using Ubuntu 11.10
> packages but I am having a hard time to get the same result running in a
> qemu/libvirt image.
>
> The first difficulty is that astribanks devices get differ
On Tue, Oct 18, 2011 at 6:21 PM, Danny Nicholas wrote:
> Hi gang,
>
> We are moving our 1.4 asterisk with DAHDI over to 10.0 with
> SIP. Everything is going nicely except that I can’t get NV_FAXDETECT to
> compile properly into 10.0. Because of this, I will have to have my
> rece
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
wrote:
> This is my complete CLI logging
>
> -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60") in
> new stack
> [Oct 4 16:19:38] WARNING[13370
On Tue, Oct 4, 2011 at 6:06 AM, Tzafrir Cohen wrote:
> On Fri, Sep 30, 2011 at 10:50:12AM -0400, Adam Moffett wrote:
>> Is there any reason not to run Asterisk on an Intel Atom board?
>
> Only if it's not strong enough. Note that "Atom" may mean some different
> things. So consider taking various
On Thu, Sep 8, 2011 at 11:21 AM, Olle E. Johansson wrote:
>
> 8 sep 2011 kl. 17:17 skrev Kevin P. Fleming:
>
>> Honestly, I'm not really sure that there is a practical solution here. ISDN
>> overlap dialing was intended for 'dumb' phones, and SIP phones aren't 'dumb'
>> :-)
>
> That's a quote th
On Thu, Sep 8, 2011 at 9:38 AM, Kevin P. Fleming wrote:
> On 09/07/2011 11:06 AM, Daniel Tryba wrote:
>
>> The aim of the quest for overlap dialing is to let the user enter a
>> number at their own pace but immediatly dial when all digits are
>> received (just like plain old ISDN does). My trunk i
On Wed, Sep 7, 2011 at 10:28 AM, Olle E. Johansson wrote:
>
> 7 sep 2011 kl. 16:20 skrev Andrew Latham:
>
>>
>>
>> On Wednesday, September 7, 2011, Olle E. Johansson wrote:
>>
>> 7 sep 2011 kl. 15:59 skrev Daniel Tryba:
>>
>> > Looking at the
On Wednesday, September 7, 2011, Olle E. Johansson wrote:
>
> 7 sep 2011 kl. 15:59 skrev Daniel Tryba:
>
> > Looking at the history of the list I don't expect any answer but lets
> > try anyway:
> >
> > Does anybody use overlap dialing from SIP devices to asterisk? Does
> > anybody have a working
On Wed, Aug 31, 2011 at 11:49 AM, Carlos Chavez wrote:
> On Wed, 2011-08-31 at 17:03 +0200, Marco Signorini wrote:
>> Hi.
>>
>> I was following this thread. We normally use Patton SmartNode SN4112
>> series to interface to FXO ports. But I'm looking for something
>> different for a future setup.
>
On Thu, Aug 25, 2011 at 4:28 PM, Jayson Rowe wrote:
>>On Thu, Aug 25, 2011 at 4:00 PM, Andrew Latham wrote:
>>
>>
>> Maybe
>> http://svn.asterisk.org/svn/asterisk/trunk/configs/cli_aliases.conf.sample
>>
>> --
>> ~ Andrew "lathama" L
On Thu, Aug 25, 2011 at 3:58 PM, Jayson Rowe wrote:
> Good Afternoon,
> I have an Asterisk box that is acting like it is passing "core show"
> before every command I type. For example, if I type sip, I will get
> "No such command 'sip' (type 'core show help sip' for other possible
> commands).
>
>
On Thu, Aug 25, 2011 at 9:26 AM, Catalin S. wrote:
> Hello,
> I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in
> sip.conf at
> [general] section the following options:
> transport=tcp
> tcpenable=yes
> tcpbindaddr=0.0.0.0
> but after all that changes i still not see tcp port r
On Wed, Aug 17, 2011 at 6:35 PM, Mike Diehl wrote:
> On Wednesday 17 August 2011 4:12:21 pm Doug Lytle wrote:
>> Mike Diehl wrote:
>> > Any other ideas?
>>
>> They should be writing out logs to your ftp server (If your provisioning
>> them that way).
>
> At the moment, my web server isn't capable
On Wed, Aug 17, 2011 at 6:01 PM, Mike Diehl wrote:
> Hi all,
>
> I've got a customer with 10 Polycom 335's and the latest(ish) firmware. For
> the most part, things are working well.
>
> However, about once a day, a given phone will just reboot. They don't do it
> all at once, and they don't do
On Thu, Aug 11, 2011 at 3:04 PM, Danny Nicholas wrote:
> Hello list,
>
> I presently use the 1.4 releases because I enjoy sleeping
> at night. I understand that 1.4 reaches end-of-life in a little over 8
> months (https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions). I
On Thu, Aug 4, 2011 at 10:36 AM, Agustina Berretta
wrote:
> Hello folks!
>
> How can I be sure a module was burned by high tension?
>
> I installed the module, configured it using dahdi_genconf -vv
>
> but when I type: asterisk -rx "dahdi show channels" I don´t see the module.
>
> Thanks a lot
ca
On Wed, Jul 27, 2011 at 9:44 AM, Claude Hayn wrote:
> We are frequently losing power during lightning storms. (Yes we have UPS,
> but often by the time power comes back up the UPS has run out of juice)
>
> We are using Asterisk with a T1/PRI card as a front end connected to our
> PBX. Whenever t
On Mon, Jul 18, 2011 at 9:20 AM, Gilles wrote:
> Hello,
>
> I'd like to run Asterisk on an embedded device, where space is scarce.
> It should be able to handle calls from a VoIP provider in SIP, calls
> from the PSTN through Dahdi, and voicemail.
>
> If someone's already done this, I'd like to kn
On Fri, Jul 15, 2011 at 12:47 PM, CDR wrote:
> I need to keep out all connection from 5 countries, which originate
> most of the Denial of Service attacks. The entries are
> around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter
> way to do this by using User Tables in iptables, that
in a mini-itx case with a 'right-angle' PCI riser card so the
>> >> TDM400p
>> >> is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive.
>> >>
>> >> I'm getting a bunch of clicks and pops on all ports.
>> >>
On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards
wrote:
> I have a TDM400p with 3 fxs and 1 fxo daughter cards.
>
> It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p
> is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive.
>
> I'm getting a bunch of clicks and po
On Sun, Jul 10, 2011 at 5:22 PM, bilal ghayyad wrote:
> I mean:
>
> What are the customers (big customers I mean) that they installed Asterisk in
> their company to be as a reference?
>
> Example: Toyota, GM, Hilton, Shiraton hotel, ... etc
>
> An example of such companies, whom?
>
> Is there a l
On Fri, Jul 1, 2011 at 2:23 PM, Doug Lytle wrote:
> Danny Nicholas wrote:
>>
>> step by step how to set up a SIP trunk just by editing sip.conf
>
> You'll find that most here don't use a GUI.
>
> Doug
Doug
Many people get addicted to the users.conf and res_phoneprov for
automatic phone provisi
On Fri, Jul 1, 2011 at 1:55 PM, Danny Nicholas wrote:
> Hey gang,
>
> I’ve got a CISCO SPA3102 that I want to set up. My
> environment is not favorable for using the Asterisk GUI interface – does
> anybody have step by step how to set up a SIP trunk just by editing
> sip.conf
On Tue, Jun 28, 2011 at 4:53 PM, Bruce B wrote:
> Hi everyone,
> When doing a "sip show settings" on Asterisk 1.6.2.18, I see the following:
> Match Auth Username: No
> Allow unknown access: Yes
> Allow subscriptions: Yes
> Allow overlap dialing: Yes
> Allow promsic. redir: No
On Mon, Jun 20, 2011 at 11:47 PM, Alex Balashov
wrote:
> I nominate this for most imaginative use of Asterisk-users of 2011.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: htt
On Tue, Jun 14, 2011 at 9:44 AM, Olivier wrote:
> Hi,
>
> I'm using a two-years old installation script for the first time on a
> Squeeze (linux 2.6.32) platform.
> For an unknown reason (might be an obvious one), Dahdi can't be loaded
> anymore.
>
> 1. First of all, it seems /dev/dahdi content wa
On Sat, Jun 11, 2011 at 10:29 AM, bilal ghayyad wrote:
> Hi All;
>
> Any one can suggest a TFTP server to be installed in Fedora (same machine
> that Asterisk is installed) to be used for Cisco IP Phones to download the
> required firmware and configuration files.
>
> Thanks for the help in adva
On Wed, Jun 8, 2011 at 5:56 PM, Satish Patel wrote:
>
> It not working on iPhone. It's saying not able to make secure connection
>
> --
> Sent from my iPhone
Satish, Can you share what the SSL/TLS Cert says? Safari and mobile
platforms have a smaller list of CAs, just to make life hard for us
s
On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant wrote:
> A number of people are reporting that Safari is not working properly with
> JIRA. Use Firefox or Chrome for now.
>
> --
> Russell Bryant
> Digium, Inc. | Engineering Manager, Open Source Software
> 445 Jan Davis Drive NW - Huntsv
tware available to analyze a pcap or similar packet
>>> dump with similar features?
>
> On Tue, 17 May 2011, Andrew Latham wrote:
>
>> I think there are some, I saw one mentioned on the BACNet mailing list...
>> ahh yes http://cloudshark.org/ takes tshark and wireshark uplo
On Tue, May 17, 2011 at 1:12 PM, Steve Edwards
wrote:
> I was debugging a turnup with Global Crossing the other day and they
> presented me with a web page that displayed a 'ladder diagram' of a call
> including a ton of detail all neatly organized in tabs and links so you
> could drill down to an
On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens
wrote:
> Hello,
>
> is there some way to make Asterisk light up a certain light on an IP-phone ?
>
> Like MWI, the message waiting indicator can light up if there is voicemail.
>
> Could this light, or even other lights (like BLF-buttons) be used to
On Mon, May 9, 2011 at 4:40 PM, Justin Sherrill
wrote:
> Anyone have some recommended equipment for alerting people to calls in a
> noisy environment?
>
> I have Polycom IP550 phones set up in some really noisy environments - our
> mine hoists - and they tend to drown out the ringers. I'm using
On Sat, May 7, 2011 at 10:12 AM, Ryan Wagoner wrote:
> On Fri, May 6, 2011 at 2:52 PM, Andrew Latham wrote:
>> On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian wrote:
>>> Has anyone used this board as an Asterisk server?
>>> http://www.supermicro.com/products/motherbo
On Sat, May 7, 2011 at 3:05 AM, Vahan Yerkanian wrote:
> On 5/6/11 11:52 PM, Andrew Latham wrote:
>>
>> On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian wrote:
>>>
>>> Has anyone used this board as an Asterisk server?
>>>
>>> http://www.supermic
On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian wrote:
> Has anyone used this board as an Asterisk server?
> http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=H&IPMI=Y
>
> I'm mostly interested about the possible compatibility issues this board may
> have with the AEX800 card.
On Wed, May 4, 2011 at 12:00 PM, A J Stiles
wrote:
...
> (For my part, I'm actually surprised that nobody came up with a proper
> protocol for encapsulating the stream of zeros and ones that make up a fax
> transmission but rely on the precise timing inherent with a circuit-switched
> network, int
On Wed, May 4, 2011 at 10:12 AM, Eric Wieling wrote:
>
> Does Hylafax support T.38?
>
> The free fax works just fine with DAHDI. I've never tried to do T.38 with
> that since it seems like it would be complicated and not give me much over
> using DAHDI.
>
There is the t38modem[1] project. But
On Tue, May 3, 2011 at 5:31 PM, bilal ghayyad wrote:
> Hi All;
>
> I need to configure the SIP account so if first IP address failed then to
> send for the second IP address. How to do this?
>
> While configuring the sip account, at the host parameter, can I give two IP
> addresses separated by
On Thu, Apr 28, 2011 at 1:34 PM, satish patel wrote:
> Where did you download asterisk 1.10 or trunk ? I search and found nothing.
> could your point me there?
>
> -S
svn co http://svn.asterisk.org/svn/asterisk/trunk /usr/src/asterisk_trunk
--
~~~ Andrew "lathama" Latham lath...@gmail.com ~~~
On Wed, Apr 27, 2011 at 3:34 PM, Olle E. Johansson wrote:
> Friends,
>
> We have a discussion on asterisk-dev about the maintenance of the 1.4 branch.
> According to the release plans, support for 1.4 was scheduled to close in
> April 2011 - basically now. After that, only security patches would
No [1]
1. AsteriskNOW does have some of these services as do many
distributions like Zentyal.
On Wed, Apr 27, 2011 at 2:04 PM, Thomas Perron wrote:
> Are there any internal DHCP or DNS services built-in to the Asterisk code?
>
> --
>
>
>
> --
> ___
On Mon, Apr 25, 2011 at 11:10 AM, Paul Belanger wrote:
> On 11-04-25 10:49 AM, David Backeberg wrote:
>>
>> On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich
>> wrote:
>>>
>>> Does this ConfBridge requires a hardware timing source?
>>
>> No, and neither does MeetMe with modern DAHDI.
>>
> This is
On Mon, Apr 25, 2011 at 10:27 AM, David Backeberg wrote:
> On Mon, Apr 25, 2011 at 9:38 AM, David Vossel wrote:
>> I am proud to announce that after a good bit of development, community
>> feedback, testing, and >code review, the brand new ConfBridge application
>> has been officially merged in
On Mon, Apr 18, 2011 at 3:06 PM, Leif Madsen
wrote:
> On 11-04-18 02:47 PM, Jerry Geis wrote:
>> When I do "core show channels concise" over the AMI interface
>> how do I specify that I want to see the "actual" channel number like
>> DAHDI/4/xxx
>> where 4 is the actual channel.
>>
>> RIght no
On Wed, Apr 13, 2011 at 10:50 AM, satish patel wrote:
> Hi Guys!
>
> I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it
> could handle in production so following is my senario.
>
> [sipp_client]---[Asterisk][sipp_server]
>
> sipp_client
> ./sipp -
On Wed, Mar 30, 2011 at 9:38 AM, vip killa wrote:
> so does anyone use fail2ban w/ asterisk or most people use sshguard?
Vip, the overall message is that it takes layers of
settings/configurations to "secure" an installation.
Simple Guide
1. alwaysauthreject = yes in
http://svn.asterisk.org/svn/
On Tue, Mar 29, 2011 at 2:34 PM, Sherwood McGowan
wrote:
> On 3/29/2011 12:25 PM, Steve Edwards wrote:
>>> On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
>>
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your database's table
On Tue, Mar 29, 2011 at 11:05 AM, Dean Hoover wrote:
> I have an Asterisk server running 1.6.2.13, where I can't seem to get
> the increased logging to save to the /var/log/asterisk/messages file.
> I have tried using the standard "core set debug 10" and "core set
> verbose 10", as well as specifi
On Mon, Mar 28, 2011 at 9:20 AM, vip killa wrote:
> Is anyone using asterisk with fail2ban? I have it working except it takes
> way more break-in attempts than what is set in "maxretry" in jail.conf
> For example, I get an email saying:
> "The IP 199.204.45.19 has just been banned by Fail2Ban afte
On Fri, Mar 25, 2011 at 7:05 PM, Carlos Chavez wrote:
> Can anyone recommend some White Papers or Success Cases that we can use
> to ease the mind of a customer that has not heard much about Asterisk? All
> they know is Avaya at this point.
>
> --
> Carlos Chavez
> Director de Tecnología
> Te
On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini
wrote:
>
>> Is there a problem having 2 telcos on the same PRI card?
>
> I think you go with one master timer as the Telco. Then the other spans are
> secondary, tertiary, quaternary timers.
>
> Adrian
Adrian
This only works when all the provide
On Fri, Mar 18, 2011 at 3:15 PM, Tiago Geada wrote:
> Just a follow up with a bit more information
>
> asterisk*CLI> module show like timing
> Module Description Use
> Count
> res_timing_pthread.so pthread Timing Interface
On Wed, Mar 16, 2011 at 10:50 AM, Jonas Kellens
wrote:
> is it possible to extract the Remote-Party-ID from an incoming call in the
> dialplan ? Is there some kind of function for this ?
>
> Kind regards,
> Jonas.
1.8 Documentation on Connected Line update. Works like magic.
https://wiki.asteris
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
--
~~~ Andrew "lathama" Latham lath...@gmail.com ~~~
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live int
On Wed, Mar 9, 2011 at 12:53 PM, Darrick Hartman (lists)
wrote:
> On 03/09/2011 02:57 AM, Dan Journo wrote:
>>
>> > could anybody suggest a usable doorphone and magnetic door opener
>>
>> > "hardphone" system for me, please? Of course should be connectable to
>>
>> > asterisk. I am in the EU, s
On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna
wrote:
> Hi every one,
> I am doing some experiments on asterisk server
> performance.. How can we know server performance? can any one explain me
> plz
> I have 2 doubts regarding the asterisk server performance
On Fri, Feb 25, 2011 at 9:49 AM, Leif Madsen
wrote:
> On 11-02-24 08:56 PM, Andrew Latham wrote:
>>
>> And I go back to triple check and compare revision numbers... You are
>> 100% correct, the revision numbers in our local repository are wrong,
>> someone pushed t
On Tue, Feb 22, 2011 at 12:16 PM, Ishfaq Malik wrote:
> Has this issue been fixed in this release of 1.8 (or even in the
> previous 1.8.2.3)?
>
> https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
>
> Thanks
>
> Ish
Ishfaq, I spoke to soon and was looking at the wrong checkout.
On Thu, Feb 24, 2011 at 7:43 PM, Leif Madsen
wrote:
> On 11-02-24 04:08 PM, Andrew Latham wrote:
>>
>> There are many updates in 1.8.2.4 that may fix your issue. If you are
>> running any version of 1.8 it should be a quick update.
>
> I wouldn't say "ma
On Thu, Feb 24, 2011 at 6:05 PM, Bryant Zimmerman wrote:
> I had an issue today where receive_fax caused an asterisk switch to crash.
> The switch is 1.8.2.3 version. The call was coming from a fax machine. The
> call started receive_fax answered and then asterisk stopped responding. I
> was able
1 - 100 of 409 matches
Mail list logo
