On Sat, Feb 11, 2012 at 8:03 AM, asterisk jobs wrote:
> Hi everyone,
>
> Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about
> 5000 numbers and then put the call to agents right away and pull up the CRM
> based on the number dialed. So, I am going to be doing some PHP+Ajax
On Thu, Jan 26, 2012 at 7:36 PM, Steve Edwards
wrote:
> The OP was using MIXMONITOR_EXEC (although I wonder about the '&&' syntax)
> so he doesn't need to explicitly execute (via system()) his commands.
Wow. Never knew that was possible. I still don't like the syntax, but
good to know.
For optim
On Thu, Jan 26, 2012 at 7:18 PM, David Backeberg wrote:
> shebang /path/to/bash
>
> PATH=$1
> lame --arguments $1.wav $1.mp3
> if [ -f {$1}.mp3 ] ; then
> rm {$1}.wav
And my silly code sample hasn't been debugged, and I can spot one
glaring bug, and another less importan
On Wed, Jan 25, 2012 at 10:29 AM, Faraj Khasib wrote:
> Hello Guys,
> I am trying to convert files that are .wac to mp3 after mixmonitor command is
> called but it doesnt execute the command, I tried the command in terminal it
> worked, any help please ... below is my dial plan
> exten=6500,n,Se
On Tue, Jan 10, 2012 at 6:00 PM, Christopher David Howie
wrote:
> I've been up and down this issue for a few hours and I cannot for the
> life of me determine why simply defining a peer causes Asterisk to offer
> telephone-event. I have tried specifying dtmfmode=rfc2833 or
> dtmfmode=auto in [glo
On Wed, Jan 4, 2012 at 4:45 PM, Asterisk Development Team
wrote:
>The Asterisk Development Team is pleased to announce the first
>release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0.
>2.6.0 is a feature release which:
>
> wct4xxp: Expose serial number in dahdi_device and kernel log.
This i
On Thu, Jan 5, 2012 at 8:05 AM, Steve Underwood wrote:
> No PAP2 or PAP2T supports T.38, even though many people will swear that they
> do. For a little while there was some beta code for the PAP2T with badly
> broken T.38 support. Perhaps this is where the "legend of T.38 on a PAP2T"
> started. O
On Wed, Dec 28, 2011 at 4:10 PM, Danny Nicholas wrote:
> Can somebody point me to an explanation from Kevin or Tzafir or someone else
> "up the food chain" explaining the differences/benefits of 1.6/1.8 vs
> 1.4/10.0?
What's the difference between a car released in 2006 versus a car
released in 2
On Fri, Nov 18, 2011 at 2:23 PM, Sazzad wrote:
> Hi,
> I have to use asterisk with some dedicated DSP chips, which will do the
> expensive G729 CODEC computing, so that the server processor has minimum
> load. I was informed, I've to use GPAK to implement this. So far I've
I had never heard of GP
On Thu, Nov 10, 2011 at 12:24 PM, Leif Madsen
wrote:
> On 11-11-10 12:12 PM, Danny Nicholas wrote:
>> Yeah! My boss will be much happier having a system that doesn't have the
>> -tail on it.
>
> I hear this kind of statement every once in a while, which makes absolutely no
> sense to me. If you'r
On Thu, Oct 27, 2011 at 11:53 AM, Mike wrote:
> I am trying to record a MeetMe conference, and this is what is relevant in
> the 1.8 manual:
>
>
>
> r - Record conference (records as MEETME_RECORDINGFILE using format
> MEETME_RECORDINGFORMAT. Default filename is
> meetme-conf-rec-${CONFNO}-${UNIQU
On Sun, Oct 23, 2011 at 3:16 PM, Tzafrir Cohen wrote:
> On Wed, Oct 19, 2011 at 10:11:08AM -0400, David Backeberg wrote:
>
>> If you use DAHDI, you need to change ownership of /dev/dahdi/* to the
>> non-root owner. I ended up rolling that into the init script for
>> dahdi
On Wed, Oct 19, 2011 at 7:19 AM, Torbjörn Abrahamsson
wrote:
> Thank you, I actually found the asterisk.conf settings after sending the
> mail. So next question is which folders/files do I need to change ownership
> of to make it work?
>
>
>
> /etc/asterisk
>
> /var/lib/asterisk
>
> /usr/lib/aster
On Mon, Sep 12, 2011 at 11:19 AM, Tarek Sawah wrote:
> i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed
> at 100 concurrent calls.
> please advise?
Nobody will know why your asterisk crashed unless you follow the
instructions here:
https://wiki.asterisk.org/wiki/disp
On Tue, Sep 6, 2011 at 10:36 PM, Leif Madsen
wrote:
> However you could select/deselect modules using menuselect if you wanted to
> automate the process. It's documented over here:
>
> http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id293439
Yeah, menuselect has been
On Tue, Sep 6, 2011 at 4:28 PM, Kevin P. Fleming wrote:
> This is a bug in the configure script, but in the meantime, you should be
> able to use "--without-pwlib" to avoid it, as long as you aren't trying to
> build chan_h323.
Thanks much.
I was trying
./configure --disable-chan_ooh323
and th
I'm having annoying errors trying to get configure working.
tar xvzf /usr/local/src/asterisk-1.8.6.0.tar.gz
cd asterisk-1.8.6.0
./configure
I get complaints related to pwlib / ptlib...
checking for openr2_chan_new in -lopenr2... no
checking /root/pwlib/include/ptlib.h usability... no
checking /r
read the 1.6 README and the 1.8 README.
If you're using SIP you should expect changes with account
authentication, faxing, output regarding channel status and
performance.
I think that version of 1.4 is late enough you would already be on
DAHDI for hardware devices. If not, you need to convert to
That debug looks cool but I have no idea what it means.
If you are using T.38, turn it off, and do audio fax, recorded with MixMonitor.
When you can hear the audio of the fax hopefully you will be able to
tell what's going on, and if you're lucky it's something specific to
the particular kind of
On Mon, Jul 11, 2011 at 5:29 PM, Steve Edwards
wrote:
> Many times, I've made the statement that you can execute hundreds of AGIs
> written in C in the time it takes to load an interpreter and parse a script
> written in PHP or Perl.
I've truly enjoyed this thread. And while startup time is certa
On Mon, Jun 27, 2011 at 9:06 AM, Michael wrote:
> Hi Kevin,
>
> Controlling it through the sip.conf peers is sufficient for us for this case
> (because this particular provider doesn't support T.38 at all), but I think
> it would be a good idea to add the option to enable/disable T.38 from the
> d
On Fri, Jun 24, 2011 at 4:55 PM, Hose wrote:
> Can anyone recommend some kind of virtual t.38 fax software? I'd like
> to test/debug some of the t.38 stuff, but it'd be much easier if I had a
> software client that could just generate the faxes from a workstation,
> rather than having to sit with
On Thu, May 26, 2011 at 9:09 AM, Ishfaq Malik wrote:
> Hi
>
> Does anyone know if there are any free UK accented English sounds packs?
Which UK accent?
I'd pay a small bit of money just to hear a Liverpool accent set of voices.
You can do some Scottish voices with Festival and the "Alan" voice.
On Thu, May 5, 2011 at 1:43 PM, vip killa wrote:
> The majority of open source projects out are NOT run by commercial
> institutions...
Postfix kicks butt. But only because IBM paid for development, for a
long number of years, and because they hired somebody who had a really
good idea how to impr
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, into so
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.
> Will I be able to use this on any virtual server without having the need
> special changes to
> the VM setup?
Define 'any'? If
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 into Asterisk >Trunk!!!
> http://svnview.digium.com/svn/asterisk
On Fri, Apr 1, 2011 at 7:04 AM, Khaled W. Chehab wrote:
> 1-Is there a way to export fax tiff file image from .pcap captured file .
Maybe, but I can't think of how. If you can somehow invert the pcap
file back into packets and reproduce the fax traffic, then maybe.
> In other words i am trying t
2011/2/3 Marcello Colucci (SIRIO Informatica s.a.s.)
:
> Hi, I have asterisk 1.6.2.6 on a Debian Lenny system.
> When I try to send a fax in T.38 mode I receive this error
>
> ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel
> 'SIP/eutelia-sirio-out-' is in an unsupported T.38 negotiat
On Tue, Jan 25, 2011 at 7:01 PM, Bryant Zimmerman wrote:
> Ok If I set t38pt_udptl = no on the trunk the fax comes in t.30 but I can't
> make t.38 work I keep getting the following error "Disconnected after
> permitted retries" Any ideas on this?
So you're saying if you turn off t38 in sip.conf
On Tue, Jan 25, 2011 at 1:45 PM, Bryant Zimmerman wrote:
> Do you know how to force off T.38 in res_fax?
it's in sip.conf
take a look for
t38pt_udptl=yes
change it to no
> reload sip
on your console
that should force it to either fail entirely or do audio passthrough.
--
_
On Tue, Jan 25, 2011 at 9:34 AM, Bryant Zimmerman wrote:
> On 01/24/2011 2:54PM Bryant Zimmerman wrote
> The attached file was too large so I am putting in a link to the file. It is
> a virus free text file.
You failed to mention earlier that this is T.38.
Turn off T.38 and see if it's still br
On Mon, Jan 24, 2011 at 4:51 PM, Steve Edwards
wrote:
> We know the problem exists -- the boss just installed U-verse at his house
> :)
> It works fine from cell and copper, just not from U-verse and their ilk.
Well, I would say more data samples are needed then. It could
certainly be the boss's
On Mon, Jan 24, 2011 at 3:37 PM, Steve Edwards
wrote:
> One of my clients is complaining that their customers that use U-verse (and
> other cable providers) for telephone service cannot enter credit card
> numbers reliably.
>
> The issue not all digits are received in my dialplan.
>
> The calls co
On Mon, Jan 24, 2011 at 2:53 PM, Bryant Zimmerman wrote:
> I am testing out inbound faxing using res_fax and res_fax_spandsp.so
>
> My system answers the call but then sets there on the ReseiveFax line then
> comes back with an error that it exceeded the maximum retries.
> How would I go about deb
On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepal wrote:
> I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can
> send recieve faxes from both boxes fine to and from pstn. But the faxing
> between 1.6 and 1.4 extensions does fail. Any ideas please ?
You don't say what's between
On Thu, Jan 20, 2011 at 10:00 AM, Flavio Miranda
wrote:
> Hi all,
> I realize that the application Receivefax can't handle with more than one
> fax at the same time. In a environment with a lot of fax, some caller get
> the signal but the operation can't be completed.
> Is there a way to send
On Wed, Jan 5, 2011 at 6:59 PM, Myles Wakeham wrote:
> For some reason our Asterisk box is doing something really unusual following
> applying a routine update to CentOS 5 on Monday.
>
> We have Asterisk 1.4.2 and its been working great for years. But now when
> the phone system receives an inc
On Mon, Dec 20, 2010 at 5:02 PM, Bryant Zimmerman wrote:
> I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql
> table for CDR's today there are no entries since the update.
> I have rebuilt and re-installed and re-started asterisk still no CDR's
> flowing to mysql. I did n
On Wed, Dec 8, 2010 at 10:17 AM, Gilles wrote:
> On Wed, 8 Dec 2010 09:33:22 -0500, David Backeberg
> wrote:
>>* pay somebody else to do it in the form of appliance and lose most
>>control versus do it yourself and have total control but also the
>>chance to screw up.
On Wed, Dec 8, 2010 at 9:06 AM, Gilles wrote:
> Hello
>
> I need to find a recent and neutral comparison of the major products
> available to connect an Asterisk server to the telephone network,
> whether ISDN (BRI) or PSTN, and through a PCI card or some external
> box. I'm told there are
On Tue, Nov 30, 2010 at 7:34 PM, Duane Larson wrote:
> I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant
> High-Availability. I was wondering if it's possible for Asterisk to also
> use multiple database servers for Realtime? Currently with Realtime I am
> only able to
On Tue, Nov 23, 2010 at 8:25 AM, voip crazy wrote:
> Hello,
>
> I want to analyze the asterisk logs files, looking for all kind of
> errors, ¿Anyboby knows any asterisk logs analyzer?
You're only going to have the logs for what you create logs for.
I create custom logs for the custom things I ne
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub wrote:
> Sorry,
> what I meant was:
> server*CLI> remove extension (hit tab)
> segfault..
> 1.4.22
> It could be an extension name Where is the error trapping if this is the
> case.. Who writes this shit?
If you remove an extension that is being used
2010/11/25 Захаров Антон :
> Hello everyone.
>
> I have a timing slips errors and I can't understand what source of the
> problem is.
> My installation has 2 digium cards: TE420 and TE220 cards in one server.
> There are 3 spans (E1) to PSTN and 3 spans to internal PBS stations -
> normal installat
On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius
wrote:
> Hi All,
> We have a requirement to record over 60 simultaneous calls. Our recording
> facilities are implemented using Monitor() over AMI. The thing we have
> noticed that making 60 simultaneous call recordings using wav CPU load is
> s
On Mon, Nov 15, 2010 at 8:30 AM, Richard Kenner wrote:
> It's kind of low for me. How does one control that volume?
I've never heard of a way to control that volume.
You can tweak after-the-fact with sox, or you can crank up your
soundcard / amplification on playback.
--
_
On Sun, Nov 7, 2010 at 1:29 PM, Cary Fitch wrote:
> But can anyone contribute some practical knowledge of systems that take in
> channel bank T1s or DS3s from "far away", and process the calls?
Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a
channelized DS3 into PRIs.
> Not
On Wed, Oct 20, 2010 at 10:35 AM, VoIP Question wrote:
> Another question: Is there (expect for the admin guide that we didn't
> succeed to understand the example in) an example somewhere for ReceiveFax
> full extensions.conf diaplan? We would like to allocate one of the
> extensions that our SIP
On Tue, Oct 19, 2010 at 1:01 PM, VoIP Question wrote:
> Digium claims that their FFA is the best and most compatible solution and
> they give one channel for free, but do not provide support for those that do
> not buy more channels, but why buy more channels if the free/test one
> doesn't work?
On Tue, Oct 19, 2010 at 11:48 AM, VoIP Question wrote:
> The whole point (as I specified in the header and initial message) is the
> attempt to use "Fax for Asterisk" to send the message.
Asterisk can handle audio passthrough faxing. I'm talking audio faxing
over SIP. You compile against this thi
On Tue, Oct 19, 2010 at 10:23 AM, marvin horst wrote:
> How did the setup work as far as extensions on the Inter-Tel system
> contacting extensions on the asterisk system?
It worked, I dare say, flawlessly. Well, as flawlessly as Inter-Tel
worked. Still had to watch out for line error counters, a
On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question wrote:
> It's set to yes for this peer.
>
> also t38pt_udptl is set to yes.
>
> :(
You don't say anything about what you're trying to send / receive against.
Here's how you should troubleshoot:
* start with a 'real fax machine' if you have one, on
On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question wrote:
> Hello,
>
> I'm trying to send a tif file, using Fax for Asterisk and the call is
> executed, but when I get the reINVITE with T.38 data, the local server
> doesn't recognize that we have this capability and sends a 488 message.
http://www
So I'm in a situation where I want to consolidate cdr logs. My general
idea is to use cdr_mysql for this.
I know I can do things like
Set(CDR(userfield)=hostname)
And I can hardcode the hostname for the dialplan on each system.
But what I'd really like to do is have this dynamic, so I can use th
On Mon, Oct 11, 2010 at 6:14 PM, Daniel Knoll wrote:
> Hey,
> i forgot to ask, how can i get the user number from a caller he is in a
> conference, i don't find a variable to us this for the current channel.
> Only the command "meetme list " shows the usernumber, but i can't use
> this output.
On Wed, Oct 6, 2010 at 5:00 PM, marvin horst wrote:
> Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone
> system via a SIP trunk using the IPRC card?
I have, believe it or not, integrated Asterisk with Inter-Tel.
However, not via SIP. Run the costs.
When I did, it was w
On Thu, Sep 30, 2010 at 11:46 AM, khalid touati wrote:
> thanks for replies,
> I am using Asterisk 1.6.2.11
> and components res_fax-1.4_1.2.1-x86_64 and
> res_fax_digium-1.4_1.2.1-barcelona_64.
> (amd 64 bit machine)
> actually I am not aware that there is version which include fax.
> for rebuild
On Thu, Sep 30, 2010 at 10:51 AM, khalid touati wrote:
> Hi List,
> I did follow the procedure to install Free Fax for Asterisk successfully
> till i came accross this isssue: i can't load the fax module:
>
> pbx3*CLI> module load res_fax_digium.so
> Unable to load module res_fax_digium.so
> Comma
On Sun, Sep 26, 2010 at 10:49 PM, Govind, Mahesh (NSN - IN/Bangalore)
wrote:
> Another reason for storing in the database is to , enable some other
> apps to access the recording at some point of time .
Yeah, still not a good reason.
File systems allow multiple read streams to the same file. Per
On Fri, Sep 24, 2010 at 1:32 PM, Don Kelly wrote:
> Don sez: I don't know how to make Outlook indent. I usually top-post, but I
> don't like getting yelled at.
>
> Why do you say "Don't do that"? Is there a real reason that it would be bad?
Performance is a real reason. Multiple simultaneous writ
On Thu, Sep 23, 2010 at 11:23 PM, Govind, Mahesh (NSN - IN/Bangalore)
wrote:
> The reason is when doing a load balancing , We cannot confine the
> recording to a particular asterisk machine ( If we have more than one
> asterisk machine in the topology ).
Yes you can. You can record the file whe
On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore)
wrote:
> HI ,
>
> Is there Any way is there so that I can store my recordings directly to a
> database rather storing the same to a file .
Please, please, please tell us why you would want to do that.
--
__
On Wed, Sep 22, 2010 at 10:00 AM, Adam Moffett wrote:
> In the simplest terms I can think of, I'm going to describe what I want to
> do and I want to know if it's possible in the current version of asterisk.
>
> Can I take a T38 call from an ATA, convert that back to analog and have
> asterisk scr
On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria wrote:
> Now I have no previous experience with Cisco systems and don't want to screw
> up anything. Are they much different than Asterisk based systems?
sometimes. Cisco supports "SIP", but depending on the product,
asterisk inter-networking with
On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria wrote:
> Now I have no previous experience with Cisco systems and don't want to screw
> up anything. Are they much different than Asterisk based systems? I guess
> the underlying VoIP technology is the same for both the systems so it
> shouldn't be
On Mon, Sep 13, 2010 at 4:33 PM, Stanislav Korsei wrote:
> Hello!
> I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
> When i try to receive fax I get:
> [Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
> 'SIP/crocus-ua-0004' refused to negotiate T
On Wed, Sep 8, 2010 at 4:18 PM, Stanislav Korsei wrote:
> Can you recommend any specific solution to this problem or way to install
> app_fax?
Not without specific debugging about what problems you're seeing. You
get a lot of information when faxes succeed or fail. Try a fax and
paste in the debu
On Fri, Sep 3, 2010 at 11:50 AM, dave george wrote:
> The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk
> SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the
> PSTN.
You don't say the percentage that are failing. However, people who
have worked with S
On Tue, Aug 24, 2010 at 9:05 AM, Ron wrote:
> hi all,
>
> i recently subscribe for an isdn and terminate it on a 3825 router.
>
> i used it as a sip trunk for my asterisk. i'm a newbie when it comes to
> ISDN. and i've been experiencing some issues:
>
> 1. Call Hangup:
>
> When hangup is initiated
On Sat, Aug 21, 2010 at 10:49 PM, Duncan Turnbull wrote:
> Voice recognition is a pain for people with accents and poor lines and when
Everybody has an accent. Some people live in a place where the people
they talk to sound like themselves, so they forget that fact.
Of course, this is a huge pro
On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr wrote:
> We gave the phone a static IP address and pointed it to the configuration
> server on the remote end that has the CFG files for it. The phone starts
> up, downloads SIP and the “new application” and otherwise seems to be
> booting normally. Th
2010/8/13 Lyle McKarns :
> Does anyone have any feelings one way or the other about running Asterisk on
> AMD vs running Asterisk on Intel?
Only political feelings. I want to support AMD so there's at least
some token competition for Intel.
Both companies make nice 6-core processors with a lot of
On Fri, Aug 13, 2010 at 11:43 AM, Eric Merkel (Mail Lists)
wrote:
>
>
> I am looking to build a small PBX for an office that has 3 incoming analog
> lines and less than 10 extensions.
For that small of an installation you might prefer an asterisk
appliance. You can review the archives, or ask for
On Wed, Aug 11, 2010 at 11:24 AM, Kent Varmedal wrote:
> We need to upgrade this PBX for it to work with SIP, it is at the moment
> using ISDN. And those who delivered it and do the
> support/reconfiguration is paid by the hour. We don't have any control
> over it our self, so when it is changed i
On Wed, Aug 11, 2010 at 4:36 AM, Tino wrote:
> Is it possible to install Asterisk on Vmware(centos) from source. Is there
> any difference or disadvantage for this compared to asterisk running on
> physical machine.
This has come up repeatedly on the list.
Basically, the less you use it, the les
On Wed, Aug 11, 2010 at 10:12 AM, Kent Varmedal wrote:
> I'm trying to set up an "old" PBX (that supports SIP) to go through our
> new Asterisk server, so that our old phones can be used still for some
> time.
>
> How can I set up Asterisk to deliver a trunk sip connection that our old
> PBX can c
On Mon, Aug 9, 2010 at 4:36 AM, Zhang Shukun wrote:
> hi, group
> there are two module can used for meeting. MeetMe and
> Conference(which is a plugin)
>
> My question is :
>
> which is better for large conference(maybe above 100 people in a meeting)?
There's at least one more choice, which i
On Wed, Aug 4, 2010 at 11:57 AM, Alejandro Cabrera Obed
wrote:
> Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
> four HD's available, using CentOS as the OS.
>
> What's the best RAID type recommendation ??? RAID 1 or RAID 5 ???
Not really an asterisk question. Asterisk wil
On Thu, Jul 29, 2010 at 4:15 AM, zeynep yildirim wrote:
> Hi All,
>
> I upgraded 7970 from SCCP to SIP. But the phone isn't registering.
> Have you got any working XML file for 7970 phones.
Isn't registering with what?
If you're registering that with CallManager, you have to change the
phone con
On Thu, Jul 29, 2010 at 5:04 PM, wrote:
> Do you know if it is possible to interconnect 1.6 with Microsoft Office
> Communications Server 2007 and use the Office
> Communicator as a softclient for telephone calls and the Communicator for
> Instant Messaging? I believe you can set up a mediation
>
So historically I've done one of two things on systems where I've
needed to use MeetMe
* used a real Digium card, and I've only ever used a TE400 or a TE420
for that purpose, and I know they have the timing chip
* used dahdi_dummy, which works well with light load, but I had it
running on a very o
On Thu, Jul 29, 2010 at 7:22 AM, Nguyen Quang Tri wrote:
> Hello,
>
> i have Cisco Unified Communications Manager with 10 ip phone,i dont buy
> license IVR of Cisco Unified Communications Manager. Can i use feature IVR
> on Asterisk connect with Cisco Unified Communications Manager.
Yes, and no.
On Tue, Jul 27, 2010 at 6:08 PM, bruce bruce wrote:
> :-) I knew someone would bring up FreePBX. I have FreePBX installed and it's
> not good for Queues at all. It's using the reporting tool from Areski and
One of the several things you asked for was GUI for cdr database logs.
FreePBX is good fo
On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce wrote:
> I seem to not be able to find any good open source Asterisk Queue Analyzer
> and Asterisk Log Analyzer on the web.
google 'freepbx'
It does some of what you want. For the rest of what you want, strongly
consider paying a professional consult
On Fri, Jul 23, 2010 at 8:46 AM, Matt wrote:
> It's not necessarily this simple. There is an approximately 50-75foot cable
> run through ceilings and walls (CAT5) to the location where the phones will
> be. At the phone location there is no power.
You always have options. You just have to decid
On Thu, Jul 22, 2010 at 2:46 PM, Matt wrote:
> I've got an interesting situation where I have one cable run from the feed
> area to the service area. I have three devices that I need to power at the
> service area. Is anyone aware of a device that will take the POE from the
> cable run and then
On Sat, Jul 17, 2010 at 6:52 PM, David Shauger wrote:
> Can anyone provide the settings in Audacity to create a proper wav file
> without having to do additional conversion in the cli? Has to be a way to do
> this with less steps.
If your goal is to 'minimize steps', you should do a batch on the
On Sat, Jun 26, 2010 at 2:09 PM, Eyal Goltzman wrote:
> Hello,
>
> After installing and learning Asterisk I found myself with a need for a
> minimal set of empty configuration files with only the "must have" stuff in
> order to setup a SIP only machine, is there a place to find it?
Depends on how
On Fri, Jun 25, 2010 at 11:00 AM, Cary Fitch wrote:
> I see some talking about TNTs in this forum. Those are 672 lines or in some
> versions double that, what is used behind them to do the processing, etc.
So a channelized DS3 is roughly 28*23 channels in US if you do one
D-channel per PRI (othe
On Thu, Jun 24, 2010 at 11:24 PM, Cary Fitch wrote:
> But, we have an opportunity to get into a big time telecom activity.
>
> It would have 2000 to 30,000 user lines per city, and we would like to have
> those brought back to a central location for control and because transport
> can be more econ
On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir wrote:
> This is a really rookie question: when should i use TE110P ISDN PRI Card?
>
> --
> Necati DEMİR
When you have a single PRI / BRI line you wish to terminate into an
asterisk system.
--
On Wed, Jun 16, 2010 at 11:50 AM, Tilghman Lesher wrote:
> On Wednesday 16 June 2010 08:21:17 David Backeberg wrote:
>> I know if I do not do an Answer() that the call is not yet picked up.
>> However, if I do a HangUp(), is that functionally equivalent? Can you
>> Hangup(
Hello List:
I'm working on a funny scenario, where I'm bouncing calls from a Cisco
call center into asterisk. Cisco call center has some logic that if a
customer calls in, an agent is logged into a given extension... if
Cisco sends a customer call to that extension, and there is a ring
with no ans
On Sun, Jun 13, 2010 at 2:59 PM, Vieri wrote:
> I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6
> compatible).
> I've taken a look at CAGI and QUIVR but their latest code releases date back
> to 2006.
> I've also seen a more recent project (wildpbx) dated 2009:
> htt
On Thu, Jun 3, 2010 at 3:56 PM, wrote:
> Hi. For several months now asterisk will mysteriously stop inserting
> records into cdr database. I am using mysql and the asterisk addons
> 1.6.2 to accomplish this. Sometimes there is a strange error about
> column names, but often there is no error,
On Thu, May 27, 2010 at 6:17 PM, Theo Band wrote:
> I used to build Asterisk from source including the zaptel-dummy module.
> Last year I decided to upgrade and use a yum repository. I hoped that
> this would be less hassle compared to manually chasing after the latest
> release, compiling etc. An
On Mon, May 24, 2010 at 9:41 AM, Kingsley Tart wrote:
> I know nothing of Trixbox but I had a problem with my own dialplan where
> there was a delay with the user selecting 0 from my IVR menu. It turned
> out that because my extensions all started with 0 (they were real phone
> numbers), asterisk
On Thu, May 27, 2010 at 4:05 AM, Theo Band wrote:
> First I noted that dahdi_dummy is no longer present in
> kmod-dahdi-linux-2.3.0.1-1.
Not exactly true.
myhost01 asterisk # lsmod | grep dahdi
dahdi_dummy 5812 0
dahdi_transcode 8968 1 wctc4xxp
dahdi_voicebus 42048
On Mon, May 24, 2010 at 7:31 AM, Marcin J. Kowalczyk
wrote:
> Medium load system (~300 simultaneous calls) crases few times a day.
> 1.6.1.19 but then upgraded to 1.6.2.7 but it's not solving issue.
>
> Any idea what can be wrong/tunned?
I've three times had unexplained crashes of asterisk 1.6.2.
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