I also get the impression that this is a well-beaten horse, as Google
queries such as this one tend to reveal:
site:lists.digium.com didx
Alex Balashov wrote:
> Never dealt with them personally, but I have been made to understand by
> everyone else I know who has in no uncertain terms th
Never dealt with them personally, but I have been made to understand by
everyone else I know who has in no uncertain terms that:
- Technical, billing and business-related issues abound.
- Support is pretty much a pointless waste of time as you suggest.
- Basic features don't work reliably.
- B
On Mon, Jan 12, 2009 at 09:55:50PM -0700, Joseph L. Casale wrote:
> I get the following error when I execute reload in the cli on one of my
> boxes with a TDM400 card w/ one FXO port:
>
> WARNING[26444]: chan_dahdi.c:14313 process_dahdi: Ignoring signalling at line
> 20.
I get the following error when I execute reload in the cli on one of my
boxes with a TDM400 card w/ one FXO port:
WARNING[26444]: chan_dahdi.c:14313 process_dahdi: Ignoring signalling at line
20.
On Mon, Jan 12, 2009 at 16:15, Karl Fife wrote:
> QUESTION: Who's in the wrong:
>
> I recently saw an example of a u-law file with a metadata header on the
> file.
> The asterisk playback function 'PLAYED' the ascii header values as if they
> were audio data, creating an audible 'click'.
>
> Afte
Jerry Geis wrote:
Jerry Geis wrote:
I just did an SVN check out and the fix for bug 14153 was not
included in the SVN checkout.
Is there something special I need to issue in the SVN checkout to get
it?
Jerry
I did not include the command I used.
svn checkout http://svn.digium.com/svn/asteri
those cards don't terminate faxes directly; that is, they aren't fax
modems. You can redirect the call to a fake fax with hylafax and
asterisk.
2009/1/12 Gregory Malsack :
> Hello All,
>
>
>
> I have a need to connect an analog device to an asterisk server. The analog
> device has 4 analog lines g
I assume most people here know what a joke DIDX is -- but in case you
didn't already know, please avoid these people.
Basic features of their service don't work, their tech support
refuses/drags their feet to fix them for a month and if you post
publicly about them, they terminate your service.
I
I have used the xorcom usb units for fax a few times, and they work
pretty well.
PaulH
Gregory Malsack wrote:
>
> Hello All,
>
>
>
> I have a need to connect an analog device to an asterisk server. The
> analog device has 4 analog lines going into it (it’s a fax solution).
> The fax solution
Philipp Kempgen wrote:
> Jerry Geis schrieb:
>> Jerry Geis wrote:
>>> I just did an SVN check out and the fix for bug 14153 was not included
>>> in the SVN checkout.
>>> Is there something special I need to issue in the SVN checkout to get it?
>
>> I did not include the command I used.
>> svn che
Jerry Geis schrieb:
> Jerry Geis wrote:
>> I just did an SVN check out and the fix for bug 14153 was not included
>> in the SVN checkout.
>> Is there something special I need to issue in the SVN checkout to get it?
> I did not include the command I used.
> svn checkout http://svn.digium.com/svn/a
Hello All,
I have a need to connect an analog device to an asterisk server. The analog
device has 4 analog lines going into it (it’s a fax solution). The fax solution
answers the analog call, then listens for dtmf. The dtmf code that is played
tells the fax device what email address to send
Tilghman Lesher wrote:
> On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
>
>> I think it happened when I upgraded an install to 1.2.31
>>
>> The variable CALLERIDNUM no longer works and CallerID(num) has to be
>> used.
>>
>
> I don't see why not. There has been no change whatsoe
Jerry Geis wrote:
> I just did an SVN check out and the fix for bug 14153 was not included
> in the SVN checkout.
> Is there something special I need to issue in the SVN checkout to get it?
>
> Jerry
>
I did not include the command I used.
svn checkout http://svn.digium.com/svn/asterisk/trunk aste
QUESTION: Who's in the wrong:
I recently saw an example of a u-law file with a metadata header on the file.
The asterisk playback function 'PLAYED' the ascii header values as if they were
audio data, creating an audible 'click'.
After realizing the click was coming from metadata (and fixing
On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
> I think it happened when I upgraded an install to 1.2.31
>
> The variable CALLERIDNUM no longer works and CallerID(num) has to be
> used.
I don't see why not. There has been no change whatsoever to that body of
code.
> I know the initi
2009/1/10 Robert Augustyn
> Hi,
> We have zabbix running and would love to be able to monitor our asterisk
> box with it.
> I believe that some sort of SNMP is build in 1.4+ correct?
> Where do I find more info or a how to on what is supported and how to use
> it?
> Thank you.
>
> _
On Mon, 2009-01-12 at 17:08 -0200, David fire wrote:
>
>
> 2009/1/12 Russell Brown
> Quoth Steve Murphy...
> >Date: Mon, 12 Jan 2009 08:51:03 -0700
> >
> >QUESTIONS:
> >
> >Which of the two would you see being useful to you?
>
> Ob
Not sure of the cost, but I would never make a statement that Quintum
is equivalent Grandstream in any sense of the word.
On Mon, Jan 12, 2009 at 2:46 PM, Vieri wrote:
> Does anyone know how much an 8-FXS Quintum gateway costs (the equivalent of a
> Grandstream GXW4008)?
>
> --- On Mon, 1/12/09,
David fire wrote:
>
>
> 2009/1/12 Russell Brown mailto:russ...@lls.lls.com>>
>
> Quoth Steve Murphy...
> >Date: Mon, 12 Jan 2009 08:51:03 -0700
> >
> >QUESTIONS:
> >
> >Which of the two would you see being useful to you?
>
> Obvious comment really but given LEG based CDR
Does anyone know how much an 8-FXS Quintum gateway costs (the equivalent of a
Grandstream GXW4008)?
--- On Mon, 1/12/09, Faraz Khan wrote:
> From: Faraz Khan
> Subject: Re: [asterisk-users] Asterisk/GXW410x IP Analog Gateway
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
>
I think it happened when I upgraded an install to 1.2.31
The variable CALLERIDNUM no longer works and CallerID(num) has to be
used.
I know the initial one was being depreciated, but I didn't see any
mention of it.
Steve
--
NetTek Ltd UK mob +44 7775 755503
UK +44 20 7993 2612 / US +1 310 85
2009/1/12 Russell Brown
> Quoth Steve Murphy...
> >Date: Mon, 12 Jan 2009 08:51:03 -0700
> >
> >QUESTIONS:
> >
> >Which of the two would you see being useful to you?
>
> Obvious comment really but given LEG based CDR, one can determine the
> 'Simple' data but you can't work it the other way.
>
>
Quoth Steve Murphy...
>Date: Mon, 12 Jan 2009 08:51:03 -0700
>
>QUESTIONS:
>
>Which of the two would you see being useful to you?
Obvious comment really but given LEG based CDR, one can determine the
'Simple' data but you can't work it the other way.
I'd therefore find LEG based CDR more useful a
On Mon, 2009-01-12 at 19:26 +0200, Apostolos Pantsiopoulos wrote:
> Steve Murphy wrote:
> > Hello!
...
> Hi,
>
> The specs look very promising. I think everyone
> here should be grateful for your efforts. In answer to your
> question I personally find both approaches very useful, although
> I
Rob Hillis a écrit :
> Administrator TOOTAI wrote:
>
>> [MyPeer]
>> host=xxx.xxx.xxx.139
>> deny=0.0.0.0/0.0.0.0
>> permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142
>> permit=yyy.yyy.yyy.yyy/255.255.255.255
>> On incoming calls, when the peer address is the one terminatin
Steve Murphy wrote:
Hello!
Most are probably bored seeing another letter about this,
but I've put in a fair amount work on a spec for rewriting
the CDR system in Asterisk, and I have some questions:
First, please look at what I've written so far:
svn co http://svn.digium.com/svn/asterisk/team/
I have a bit of a memory leak in Asterisk 1.2, but only when using an
analogue card (TDM400 or equivalent) and wctdm + zaptel
Asterisk eventually grows over a period of days/months (depending on how
busy it is) then crashes and burns. I've seen it with all versions of
asterisk up to 1.2.31, bu
RED is just off-hook or unavailable (at least in my shop).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of fidibus83
Sent: Monday, January 12, 2009 11:04 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] a zap
Hello,
I have a problem with zaptel. I hope you can help me.
I installed and configure zaptel.
ZAPTEL.CONF
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone = de
defaultzone=de
But the output of cat /proc/zaptel/*
Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Ca
2009/1/12 Jeff LaCoursiere
>
> On Mon, 12 Jan 2009, David fire wrote:
>
> > hi again mybe this info is usefull to solve this problem
> >
> > *box1<--->*box2<>*box3
> >
> > box2 originate 1 call to box1 and to box 3
> > using sip/box1/1
> >
> > extension 1
> > context default
> >
> > exten =>
I just did an SVN check out and the fix for bug 14153 was not included
in the SVN checkout.
Is there something special I need to issue in the SVN checkout to get it?
Jerry
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
aster
On Mon, 12 Jan 2009, David fire wrote:
> hi again mybe this info is usefull to solve this problem
>
> *box1<--->*box2<>*box3
>
> box2 originate 1 call to box1 and to box 3
> using sip/box1/1
>
> extension 1
> context default
>
> exten => 1,1,dial(sip/box3/1)
>
> box1 and box3 will exec musico
hi again mybe this info is usefull to solve this problem
*box1<--->*box2<>*box3
box2 originate 1 call to box1 and to box 3
using sip/box1/1
extension 1
context default
exten => 1,1,dial(sip/box3/1)
box1 and box3 will exec musiconhold when they answer.
2009/1/12 David fire
> hi
> i am
I was able to send the xml directly to the device like:
http://ip/admin/resync?tftp://ip_of_tftp_server/myconfig.xml
This is a great start to what I want to do, but isn't really the end goal.
I put a call in to Linksys about this so maybe they'll call me back today.
Another issue I've got with thes
Hello!
Most are probably bored seeing another letter about this,
but I've put in a fair amount work on a spec for rewriting
the CDR system in Asterisk, and I have some questions:
First, please look at what I've written so far:
svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs
and look at
On Sat, 10 Jan 2009, Tzafrir Cohen wrote:
>
> No idea, but the driver is much more aware of the specifics. So maybe
> their driver has extra debugging information for that case.
>
> For starters, have you enabled full debugging in Asterisk? Make sure you
> log 'debug' and set debug to at least 5
recompile whitout dahdi suport or correct the dahdi files so they show your
real dahdi hardware if you havent any leave them blank.
2009/1/12 nik600
> PS:
>
> asterisk is compiled with dahdi support
>
>
> On Mon, Jan 12, 2009 at 1:39 PM, nik600 wrote:
> > Hi to all.
> >
> > I'm trying to use m
hi
i am using asterisk 1.4.22
ubuntu 8.4
i have two Ethernet one for ssh and other one only for voip calls
when i start a call using originate in the manager or the cli
in the voip Ethernet i get something like 4Mbits/sec of traffic only 1 G711
call.
if i start the call using a soft phone everyt
PS:
asterisk is compiled with dahdi support
On Mon, Jan 12, 2009 at 1:39 PM, nik600 wrote:
> Hi to all.
>
> I'm trying to use meetme on asterisk 1.4.22.1.
>
> On a debian i've compiled (as i need h323 support)
>
> openh323_v1_18_0
> pwlib_v1_10_0
> dahdi-linux-2.1.0.3
> dahdi-tools-2.1.0.2
> as
Philipp Kempgen schrieb:
> Max Alex schrieb:
>
>> If i got the NOANSWER then the channel is not passing to next priority.
>> I need to pass that channel to the next priority of the context
>> [macro-mypbx] so i can set voicemail there.
>>
>> I want to know how can we set the local channel to go i
Here you go http://tinyurl.com/a7tkkz
2009/1/12 chinmay chakraborty
> Hello,
>
> I am having problems getting one xlite clients to communicate through
> asterisk. I am getting an error message:
> chan_sip.c:15593 handle_request_register: Registration from '"chinmay
> chakraborty">' failed for
Hi, All
How to enable the transfer in Asterisk 1.6?
Not found the module res_features.so.
blindxfer => #
atxfer => *2
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asterisk-users mailing list
To UNSUBSCRIBE or update option
chinmay chakraborty schrieb:
> I am having problems getting one xlite clients to communicate through
> asterisk. I am getting an error message:
> chan_sip.c:15593 handle_request_register: Registration from '"chinmay
> chakraborty">' failed
> for '10.44.32.193' - No matching peer found
>
> sip sh
Hi All,
We have already use 'g' option in that, but it is not working in my case.
Thanks,
Max Alex
Voip Developer
On Sun, Jan 11, 2009 at 1:51 AM, Philipp Kempgen
wrote:
> Max Alex schrieb:
>
> > If i got the NOANSWER then the channel is not passing to next priority.
> > I need to pass that cha
I did this a long time ago, and just based it on a PAP2T XML
configuration, with 8 lines instead of 2, and it worked fine. Sorry I
don't have any useful examples to hand anymore. Are you sure it is not
just a missing slash or angle-bracket in your source XML? Try opening
it in a browser to see if i
Hi to all.
I'm trying to use meetme on asterisk 1.4.22.1.
On a debian i've compiled (as i need h323 support)
openh323_v1_18_0
pwlib_v1_10_0
dahdi-linux-2.1.0.3
dahdi-tools-2.1.0.2
asterisk-1.4.22.1
All works fine, dahdi status is:
asterik:/data/programmi# /etc/init.d/dahdi status
### Span 1:
Hello,
I am having problems getting one xlite clients to communicate through
asterisk. I am getting an error message:
chan_sip.c:15593 handle_request_register: Registration from '"chinmay
chakraborty">' failed
for '10.44.32.193' - No matching peer found
sip show peers
Name/username
open a bug report
mailinglists schrieb:
> Hi all!
>
> Am I missing some configuration or is it simply a bug: If
> Asterisk chan_sip is configured with bindport=5062, the port is missing
> on the outgoing SIP messages contact header.
> This resulting in in-dialog messages sent to port 5060 ...
I use Asterisk 1.4.22 without problems:
Contact:
regards
klaus
mailinglists schrieb:
> Hi all!
>
> Am I missing some configuration or is it simply a bug: If
> Asterisk chan_sip is configured with bindport=5062, the port is missing
> on the outgoing SIP messages contact header.
> This result
Dave Platt schrieb:
>> Bad plan? Could quite easily turn into a DoS.
>
> If the reaction is to lock the account, I agree, it might
> leave you prone to a denial-of-service attack.
>
> A better way would be to use iptables to start dropping
> packets from the IP address(es) involved in the atta
Hi Philipp!
thanks for the detailed explanation.
Philipp Kempgen schrieb:
>
> === Amooma ===
>
> * http://www.amooma.de/asterisk/sprachbausteine/#prompts-tts
> These files are generated by our web-based text-to-speech engine.
> Pros: If you need additional custom prompts, just go to
> http://w
Well, at worst you can make use of the exec and pass_persist feature of
snmpd
that way you can build you own script that will run locally to the
asterisk box
and query using snmp
Grygoriy Dobrovolskyy a écrit :
> I wonder if the same is possible with centreon ?
> Someone is using centreon here ?
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