On 8/28/07, fateme fatah [EMAIL PROTECTED] wrote:
Hi:
I think app-conference is used where there isn't zaptel hardware,in the
other word when we use zaptel hardware we shouldn't use app-conference for
conference call sevice and we should use meetme application and load
ztdummy.Is it true?
- Original Message -
From: Jody Gugelhupf [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, August 27, 2007 3:55 PM
Subject: [asterisk-users] voip provider settings problem, please help
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but
before i
We have a similar set up. I would recommend also using SER and load
balancing so you can load balance your calls out between your asterisk
box's.
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Dear Tzafrir;
Sorry I did not understand what do you mean by:
Does it work with '-T' and 'use strict'?
Do u mean the ASTPP or the prepaid billing? Where I
have to run '-T' and 'use strict'?
You do not think that I need to do download the
prepaid billing software or it come with Asterisk?
snip
I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM
and a couple 250gig SATA drives. Totally VoIP so I cannot comment on
cards or interrupts, but so far it has been flawless.
I would like to see how many G729/ULAW conversions it could handle. How
would I go about
Hi list,
I have a terrible noise issue with Dell SC1430 + Digium TE110P. The
digium card is not sharing interrupts with any other device, as I saw in
Dell's BIOS and also with lspci -vb command.
After changing coax wire, UTP, balum, digium card ... I have found that
the problem is in Dell
Hello,
I want to create php rate script and I'm using Deadagi. But I allways get
billsec 0 , or nothing. Can you help me to solve this problem...
My extension.conf:
exten = _123,1,DeadAgi(rate.php)
exten = _123,2,hangup
And my simple test php script rate.php
#!/usr/local/bin/php -q
?php
Marc Patino Gómez wrote:
Hi list,
I have a terrible noise issue with Dell SC1430 + Digium TE110P. The
digium card is not sharing interrupts with any other device, as I saw in
Dell's BIOS and also with lspci -vb command.
After changing coax wire, UTP, balum, digium card ... I have found
Dovid B wrote:
snip
I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM
and a couple 250gig SATA drives. Totally VoIP so I cannot comment on
cards or interrupts, but so far it has been flawless.
I would like to see how many G729/ULAW conversions it could handle.
That is why I suggested Sangoma. Ask them if you can return it if it
does not fix your problem.
It is alot easier than disabling things in BIOS and hunting for the
elusive noises.
Digium would have you believe that the problem is the Dell box but if
a Sangoma card works perfectly in the same
On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
Dear Mr Gavin,
Sorry for having miss pelt your name twice... Thank you once again for your
prompt reply. Is this the correct version of the driver for Asterisk 1.2.x :
res_config_ldap-v0.7.tar.gz from the link
Hi Steve,
All my cards are Digium, I tried diferent Digium cards and I had the
same problem.
Regards,
Marc
Steve Totaro wrote:
Marc Patino Gómez wrote:
Hi list,
I have a terrible noise issue with Dell SC1430 + Digium TE110P. The
digium card is not sharing interrupts with any other
Hi Steve,
Thanks for your advice, I will order a Sangoma card and test the box. A
part from this, you know any other point to recomend Sangoma cards
versus Digium cards?
Many thanks,
Marc
Steve Totaro wrote:
That is why I suggested Sangoma. Ask them if you can return it if it
does not
Hi everyone,
I am writing a project that uses the Asterisk Manager Interface to
monitor events. I just wanted to confirm if the types of messages sent
back by the AMI are
- Event
- Response
- Status
If there are any other can anyone please point them to me or point me
to some documentation
Hi
any body have idea about asterisk call waiting with SIP if we use
asterisk1.4.5 hard phone with extensions.
i need dial plan logic for this which capable to activate deactivate such
feature.
with ATA IP phone it is possible but with normal hardphone SIP in
asterisk is it possible
I will let you judge that for yourself. I suggest that you email
someone at Sangoma sales *directly* (not a reseller) to ask about buying
the card and if it can be returned if it does not work properly.
Explain your issue with the Digium card briefly.
See how it turns out for you.
Thanks,
On 8/28/07, Giedrius Augys [EMAIL PROTECTED] wrote:
Hello,
I want to create php rate script and I'm using Deadagi. But I allways get
billsec 0 , or nothing. Can you help me to solve this problem...
It seems that there is problem with Answer(). Does it get executed
from AGI? Do you hear voice
Hi,
I thought I'd give a follow up to this discussion for the archives...
Currently I'm trunking 30 channels of g.729 traffic (no transcoding going
on, the call comes in and goes out as g.729) and it takes about 350 kbps
bandwith bidirectional.
So on average each call takes 11.5 - 12 kbps
I solve this problem. I'm not sure, but you get billsec when you use Dial
application. Using dial app , I get billsec.
2007/8/28, Atis [EMAIL PROTECTED]:
On 8/28/07, Giedrius Augys [EMAIL PROTECTED] wrote:
Hello,
I want to create php rate script and I'm using Deadagi. But I allways
get
On 8/28/07, Giedrius Augys [EMAIL PROTECTED] wrote:
I solve this problem. I'm not sure, but you get billsec when you use Dial
application. Using dial app , I get billsec.
Well, Dial usually changes ANSWER status, because you usually answer
phone that is ringing. If you won't answer, you won't
Hi,
I have a PAP2 with 2 phone ports.
When I call them everything works fine until I hung up the call. There
is about 30-40 seconds until I can call to that extension again.
Before that it gives me busy messages.
Extension config:
exten = 199,1,Dial(SIP/199,30)
exten = 199,102,Hangup
Any
Motherboard with SATA RAID1 support
That's a mulit-port SATA controller with RAID in the driver (software).
256 MB RAM
Use a little more RAM.
digium PRI/E1 card
Is there any reason you aren't using Sangoma cards?
1. If I use Software RAID, what would be the impact to my deployment? (
Dear Andrew,
Thanks for your kind responce.
Regards,
vidura.
=
Motherboard with SATA RAID1 support
That's a mulit-port SATA controller with RAID in the driver (software).
256 MB RAM
Use a little more RAM.
digium PRI/E1 card
Is there any reason you aren't
hi Anselm :)
thx for your tip, though i have qualified turned on, anyhow here are my
complete sip.conf and
extensions.conf, thx for any help :)
sip.conf
[general]
allowoverlap = yes
realm = mydomain.tld
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
tos = lowdelay
disallow = all
allow =
hi ppl :D
my configuration is as follows, i run (let's call it machine 2) debian etch 4.0
and asterisk 1.2,
i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a
grandstream/handytone 486 as a sip device, no PSTN line or anything like that
all SIP only. I have
a machine
Marc Patino Gómez wrote:
Hi Steve,
Thanks for your advice, I will order a Sangoma card and test the box. A
part from this, you know any other point to recomend Sangoma cards
versus Digium cards?
Many thanks,
Marc
5 year warranty, to name one.
Sangoma says their cards will work in
Hi John,
thanks for this usefull info
Marc
John Novack wrote:
Marc Patino Gómez wrote:
Hi Steve,
Thanks for your advice, I will order a Sangoma card and test the box. A
part from this, you know any other point to recomend Sangoma cards
versus Digium cards?
Many thanks,
Marc
On Aug 28, 2007, at 8:24 AM, Jody Gugelhupf wrote:
-- Now forwarding SIP/9083XXX-0816b208 to 'Local/
[EMAIL PROTECTED]' (thanks to
SIP/486-081d4738)
Because SIP/486 issued a 302 redirect to 247110358. Check the phone
for the forwarding setting.
/b
On 8/28/2007 at 9:30 AM, John Novack [EMAIL PROTECTED]
wrote:
Marc Patino Gómez wrote:
Hi Steve,
Thanks for your advice, I will order a Sangoma card and test the
box. A
part from this, you know any other point to recomend Sangoma cards
versus Digium cards?
Many thanks,
Marc
5
Everyone,
I will be attending Astricon in Phoenix and would like to have a
little get together to discuss Open Source Telephony and the
challenges we as developers and system integrators face. Exchange
ideas and go over some use cases and see how we can all work together
to
Hello,
I do not think that the presence bit will be crucial to our application.
Thanks for your help. I will keep you posted if I get any progress.
Thanks,
Denis
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gerald A
Sent: Monday, August 27, 2007
I've purchsed 5 g729 licenses from digium, but am a little confused
about why things are acting the way they are.
When I do show g729 I see:
0/0 encoders/decoders of 5 licensed channels are currently in use
My sip.conf starts out:
[general]
disallow=all
allow=ulaw
allow=g726
then my hosts look
Hi,
I'm trying to connect an HDL F10 device for a friend living in Brazil to
the TDM2400 on his Asterisk server.
That device should behave like a normal doorbell and it is if connected
to an analog PBX.
I connected to the TDM2400 and everything works fine except for one
thing: when the called
Joe Acquisto wrote:
On 8/28/2007 at 9:30 AM, John Novack [EMAIL PROTECTED]
wrote:
Marc Patino Gómez wrote:
Hi Steve,
Thanks for your advice, I will order a Sangoma card and test the
box. A
part from this, you know any other point to recomend Sangoma cards
and reload, strange things begin to happen. A show g729 shows this:
5/0 encoders/decoders of 5 licensed channels are currently in use
I think you have a loop of some kind. As you can see none of those call
are actually established since no decoders are in use. Try to debug and
see why
Matt wrote:
I've purchsed 5 g729 licenses from digium, but am a little confused
about why things are acting the way they are.
When I do show g729 I see:
0/0 encoders/decoders of 5 licensed channels are currently in use
My sip.conf starts out:
[general]
disallow=all
allow=ulaw
allow=g726
oohs no!
Whats up, haven't heard much out of you lately.
Chris
Brian West wrote:
Everyone,
I will be attending Astricon in Phoenix and would like to have a
little get together to discuss Open Source Telephony and the
challenges we as developers and system integrators
Howdy. I've been having trouble finding a fairly modern server that meets the
following requirements:
- Molex power connectors (don't want to use the Digium FXS power supply)
- 4 PCI slots, one full-length (TDM2400P, 2xT100P, additional NIC)
- dual power supplies
- preferably dual CPUs = 1GHz
-
Hi all,
I'm kind a New to Asterisk.But I'm a Network Administrator with 5 years of
experiance.
I want to know for an installation with 90 clients, If I don't want to have
just 1 server for it, then how is it possible to distribute it among about 3
servers.
Should I do it in a cluster (kernel
Stefano Arata wrote:
Hi, I have two asterisk with the Digium TDM400 installed on the first and
the TDM800 installed on the second. Both systems are linux Debian 4.0 whith
kernel 2.6.18 and asterisk 1.2.24.
Often the cards stop answering calls, and I can't make or receive calls; I
need to
Well does g729 have to run on both legs of a call? For instance, when
I have 5/0 and I make a call from a SIP device... I get 5/1.. I can't
hear any audio on my SIP phone, however if I call someone they can
hear me.
On 8/28/07, Andres [EMAIL PROTECTED] wrote:
and reload, strange things begin
Hi!
I have this error compiling Zaptel 1.4.4
make: *** No rule to make target `xpp/xpp_usb.ko', needed by
`install-modules'. Stop.
The Zaptel 1.2.5 compile ok.
Any ideas??
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
On Aug 28, 2007, at 10:14 AM, Seysan wrote:
Hi all,
I'm kind a New to Asterisk.But I'm a Network Administrator with 5
years of experiance.
I want to know for an installation with 90 clients, If I don't want
to have just 1 server for it, then how is it possible to distribute
it
Show channels doesn't show me the codec a call is using, unless I'm
missing it somewhere... is there someway I can find out which channels
are using what codec?
On 8/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
Matt wrote:
I've purchsed 5 g729 licenses from digium, but am a little confused
gincantalupo wrote:
Hi,
I'm trying to connect an HDL F10 device for a friend living in Brazil to
the TDM2400 on his Asterisk server.
That device should behave like a normal doorbell and it is if connected
to an analog PBX.
I connected to the TDM2400 and everything works fine except for one
Stephen Kratzer wrote:
Howdy. I've been having trouble finding a fairly modern server that meets the
following requirements:
- Molex power connectors (don't want to use the Digium FXS power supply)
- 4 PCI slots, one full-length (TDM2400P, 2xT100P, additional NIC)
- dual power supplies
-
haha you going to be there?
/b
On Aug 28, 2007, at 9:30 AM, Chris Childress wrote:
oohs no!
Whats up, haven't heard much out of you lately.
Chris
Brian West wrote:
Everyone,
I will be attending Astricon in Phoenix and would like to have a
little get together to
I just set up a t1 with 2 camma mf 911 trunks on it, and I am having a issue
with it. We can call 911 which is routed out these new trunks, and it goes
to the 911 center, but they are not getting the ANI and hence no record
found. Our LEC is Embarq, and they say they can see the call come in and
Matt wrote:
Well does g729 have to run on both legs of a call?
If the call is established and there is audio both ways then yes. If
the call has not been answered yet then you will only see 1 encoder
used. In your case somebody is using up 5 encoders and it is probably
from calls coming
On Tue, Aug 28, 2007 at 10:39:09AM -0500, Andres wrote:
For instance, when
I have 5/0 and I make a call from a SIP device... I get 5/1.. I can't
hear any audio on my SIP phone, however if I call someone they can
hear me.
That is expected since when you tried to make the call there were no
Hi Matthew,
I asked HDL some docs about their product but nothing is explained
about signalling.
I've tried reversepolarity and busydetect without success.
I know 4 things:
a) it works with analogic PBX
b) both F10 and called party phone are connected to the PBX so no telco
line is involved
c)
On Tue, Aug 28, 2007 at 12:00:47PM -0300, equis software wrote:
Hi!
I have this error compiling Zaptel 1.4.4
Any reason you don't use 1.4.5.1 ?
make: *** No rule to make target `xpp/xpp_usb.ko', needed by
`install-modules'. Stop.
The Zaptel 1.2.5 compile ok.
Any ideas??
What
(With regard to your final question)
As far as I can tell, EAGI is AGI with the extra file descriptor devoted
to the linear pcm audio stream. As such, I would assume, but have never
tested, that in order to send DTMF OUT from your AGI app, you would need
to use the AGI functions, i.e. EXEC
On Tue, Aug 28, 2007 at 09:51:41AM -0400, Joe Acquisto wrote:
When I voiced that concern to the Digium techs, they set up a thing
called screen (I think it was) to allow me to see and or interact with
their session.
gnu screen is a standard prorgram available in most distributions. I
usually
Ok,
So does someone want to explain this:
(About 5 hours after I enabled G729... reloaded... then had the
problem... then disabled G729.. and reloaded I still have):
1*CLI show g729
5/0 encoders/decoders of 5 licensed channels are currently in use
Yet, no one is using these:
iax2 show channels
My linux kernel version is v2.6.15 (Gentoo)
I think my kernel need some usb modules. At night I try to compile anwe
kernel with usb options.
I'll try to use Zaptel 1.4.5.1
On 8/28/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Aug 28, 2007 at 12:00:47PM -0300, equis software wrote:
Hi!
ram wrote:
app_meetme can use ztummy but on highload expect to use hardware source
A thing that was on my mind for quite some time now:
Would it be beneficial to have a Zaptel compatible card
in a system just as a timing source, even if it's not
connected to a PRI?
Regards,
Philipp Kempgen
On Tue, Aug 28, 2007 at 02:07:58PM -0300, equis software wrote:
My linux kernel version is v2.6.15 (Gentoo)
I think my kernel need some usb modules. At night I try to compile anwe
kernel with usb options.
I'll try to use Zaptel 1.4.5.1
So you probably don't have USB support in the kernel.
As Tzafrir stated, it will NOT work with 1.2.x.
Where is this html.conf, which README? I will update it.
I will write a brief page on setting up the *GUI for all who want to
know..
There are SOME GUI's that work with 1.2, however, I almost guarantee
none of them are client side, such as
Jay R. Ashworth wrote:
On Tue, Aug 28, 2007 at 10:39:09AM -0500, Andres wrote:
For instance, when
I have 5/0 and I make a call from a SIP device... I get 5/1.. I can't
hear any audio on my SIP phone, however if I call someone they can
hear me.
That is expected since when you tried to
Matt wrote:
Ok,
So does someone want to explain this:
(About 5 hours after I enabled G729... reloaded... then had the
problem... then disabled G729.. and reloaded I still have):
I can't explain that. I have only seen stuck encoders/decoders on very
old versions of Asterisk. I remember on
Ok, but how can I do that??
Sorry I'm new in Linux/Asterisk world!
Thanks Tzafrir
On 8/28/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Aug 28, 2007 at 02:07:58PM -0300, equis software wrote:
My linux kernel version is v2.6.15 (Gentoo)
I think my kernel need some usb modules. At night
On 7/9/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi Stefan -
What I want to accomplish:
- calls within the LAN are re-invited (RTP goes from endpoint to
endpoint)
- asterisk detects when a call is going beyond the local LAN (over the
NAT),
and then stays in the middle.
I'm
The question I always have when someone mentions distributing the load
across multiple machines is how do you handle contexts for phones on
different machines? I want all of my phones to dial into
[companyA-phones]. I have to define it in two different places (or more
depending on the number
HI all,
Has anyone succesfully installed an AVM Fritz! card on Fedora 7 using the
drivers at ATrpms recently? http://atrpms.net/dist/f7/fcpci/
I tried with a clean F7 build on my EPIA 5000 yesterday, after modifying
/etc/capi.conf (removing the coment # in front of fcpci line) I received the
Andrew Ott wrote:
ZAPATA.conf
===
;911 group
group = 2
restrictcid=yes
signalling = e911
channel = 25-26
===
...
I've tried it with either one of those ${EXTEN} which just does 911, and the
${CALLERID(ani)} both have the same result no number
Hi list,
I'm running current SpanDSP
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
with Asterisk 1.2.22 somewhat successfully.
Most Fax machines do work but I have problems with people having
Tobit FaxWare and Shamrock CapiFax.
Christian Peter wrote:
Can anybody help me with this issue. Please no switch to Hylafax
mails, because I'm very happy with SpanDSP, it integrates nicely and
works most time.
*chuckle*
Lee.
___
--Bandwidth and Colocation Provided by
We are running 1.2.6 Asterisk version.
On 8/28/07, Andres [EMAIL PROTECTED] wrote:
Matt wrote:
Ok,
So does someone want to explain this:
(About 5 hours after I enabled G729... reloaded... then had the
problem... then disabled G729.. and reloaded I still have):
I can't explain that. I
Peder @ NetworkOblivion wrote:
The question I always have when someone mentions distributing the load
across multiple machines is how do you handle contexts for phones on
different machines? I want all of my phones to dial into
[companyA-phones]. I have to define it in two different
Christian Peter wrote:
Can anybody help me with this issue. Please no switch to Hylafax
mails, because I'm very happy with SpanDSP, it integrates nicely and
It just show you how many people on this list are pleased with HylaFAX+
Doug
--
Ben Franklin quote:
Those who would give up
Matt wrote:
We are running 1.2.6 Asterisk version.
That is clearly a bug then. You should open a bug report. There are
other similar strange things like the one in:
http://bugs.digium.com/view.php?id=9526
On 8/28/07, Andres [EMAIL PROTECTED] wrote:
Matt wrote:
Ok,
So does
On Aug 28, 2007, at 3:49 PM, Doug Lytle wrote:
Christian Peter wrote:
Can anybody help me with this issue. Please no switch to Hylafax
mails, because I'm very happy with SpanDSP, it integrates nicely and
It just show you how many people on this list are pleased with
HylaFAX+
Doug
--
On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote:
Hi list,
I'm running current SpanDSP
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
with Asterisk 1.2.22 somewhat successfully.
Shouldn't you have used spandsp 0.0.3 with asterisk 1.2 ?
--
Hello all -
I'm about to deploy an asterisk server here at work. Before
deploying, I'd like to do an extended load test on the system. I
currently have T1 crossover cables connecting ports 1-2 and 3-4.
Would there be an easy way to script generating a bunch of calls
across these spans? I
Asterisk Users,
I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System
2.9.18-4-amd64. A TDM03B is installed on the Debian System.
Every time, I try to change my voicemail pin via the Sip phone, the
voicemail.conf does not get modify and I see this warning message on the
Is there anywhere that we can look into for Realtime + MySQL that you
mentioned?
or about SER?
Thanks
On 8/28/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Peder @ NetworkOblivion wrote:
The question I always have when someone mentions distributing the load
across multiple machines is
On 8/28/07, Stefan van der Eijk [EMAIL PROTECTED] wrote:
On 7/9/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi Stefan -
What I want to accomplish:
- calls within the LAN are re-invited (RTP goes from endpoint to
endpoint)
- asterisk detects when a call is going beyond the local LAN
Having calls connected for that duration is worthless testing... What
you need to do is call setup and tear down many times per second... I
recommend trying to accomplish 20-30cps at 1ms to 10ms variable
durations. That will expose any bugs quickly.
And that my friend is how you expose any
On 8/28/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Realtime + MySQL does it. That needs some extra work but
it's possible.
Or DUNDi. JR just posted a quick tutorial on getting that up and running:
ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf
--
j.
On Wed, 2007-08-29 at 00:03 +0300, Tzafrir Cohen wrote:
On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote:
Hi list,
I'm running current SpanDSP
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
with Asterisk 1.2.22 somewhat successfully.
Shouldn't you
Hello John,
I think it is not the problem with your Asterisk, it is with your Phone (IP
Phone or Softphone)
Check the dtmf format on that. I think it is set to inbound, then change it
to rfcxx.
Then it should work fine.
Regards,
AFShin
On 8/28/07, John Meksavan [EMAIL PROTECTED] wrote:
On 8/28/07, Brian West [EMAIL PROTECTED] wrote:
What exactly are your needs? I can provide you some sipp scripts
that might help you.
Brian - thanks for the reply. If you read my email, I believe I make
it fairly clear what my needs are. I have a 4-port Sangoma PRI card
installed. Crossover
Realtime and DUNDi covers all the bases.
On 8/28/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
The question I always have when someone mentions distributing the load
across multiple machines is how do you handle contexts for phones on
different machines? I want all of my phones to dial
This fails to take into account total failure of a machine. NAT
mappings and various other variables that are not covered by Dundi or
realtime... Best thing is to use OpenSER in the front then failure
isn't a huge issue.
/b
On Aug 28, 2007, at 4:40 PM, Bruce Reeves wrote:
Realtime and
We are at about 250 days of 24/7 uptime now. It would be more but we had a
long power outage and the UPS's ran out. We are using Sangoma cards though.
You can easily substitute the 2U for a 3U but I don't think you need it.
Qty 1 Supermicro SC823T-R500LP, 2U, redundant 500W ps w/ PFC, 6x1 SATA
Seysan,
I tried changing the DTMF format to RFC2833, but it did not help. Any
other suggests?
From: Seysan [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Seysan wrote:
Is there anywhere that we can look into for Realtime + MySQL that you
mentioned?
Maybe
http://www.voip-info.org/wiki/view/Asterisk+RealTime
http://www.asteriskguru.com/tutorials/realtime_pgsql.html
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied -
Your inabiity to configure a card doesn't make the competitors card
better just because you said so.
On 8/27/07, shadowym [EMAIL PROTECTED] wrote:
They know what they are doing and do a lot of it. I don't have to give an
Who is 'they'?
opinion myself. There is enough evidence all over for
The README is here: svn co
http://svn.digium.com/svn/asterisk-gui/branches/asterisknow
/Configuration
=
You may install sample configuration files by doing make samples.
Also you
will need to edit your Asterisk configuration files to enable the GUI
properly,
specifically:
1) In
While I can't say this won't work the way you have it, I CAN say it's
not the way mine is set up and it's not a way I've SEEN it ever set up.
Could it just be complaining that you've got nothing on the right side
of the = for mailbox 200?
Or could it be complaining that you don't have anything
Dear Philipp;
How can I add the verbose and debug to the consol
entry in the logger.conf to be able to take logging
about the attempt of registeration for the sip end
point?
Regards
Bilal
If secret enabled, then some endpoints can not
register (maybe due to compatibility in reading the
Erik Anderson wrote:
On 8/28/07, Brian West [EMAIL PROTECTED] wrote:
What exactly are your needs? I can provide you some sipp scripts
that might help you.
Brian - thanks for the reply. If you read my email, I believe I make
it fairly clear what my needs are. I have a 4-port
bilal ghayyad wrote:
How can I add the verbose and debug to the consol
entry in the logger.conf to be able to take logging
about the attempt of registeration for the sip end
point?
console = notice,warning,error,debug,verbose
as explained in /etc/asterisk/logger.conf
Regards,
Philipp
Erik Anderson wrote:
On 8/28/07, Brian West [EMAIL PROTECTED] wrote:
What exactly are your needs? I can provide you some sipp scripts
that might help you.
Brian - thanks for the reply. If you read my email, I believe I make
it fairly clear what my needs are. I have a 4-port
Please don't feed the trolls. ;-)
C F wrote:
Your inabiity to configure a card doesn't make the competitors card
better just because you said so.
On 8/27/07, shadowym [EMAIL PROTECTED] wrote:
They know what they are doing and do a lot of it. I don't have to give an
Who is
Mojo,
Thanks for helping me with this issue. You must have a NAME and EMAIL
address after putting in the voicemail pin.
I just migrate to Asterisk 1.4.x from 1.2.13, so I am still trying to get
use to all the new stuff in the newer version. In Asterisk 1.2.13, it is
not necessary to
Hi,
I've been avoiding investigating this issue for a while; I used to
revert to a previously compiled version of zaptel a previous kernel
(as at some point I think I stopped being able to compile the older
zaptel against the newer kernels) and all was well. However I've now
upgraded kernels
I am looking at Thirdlane's solution now. Very impressive and modest cost.
The asterisk GUI is free :]
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Steve,
That is http.conf, not html.conf.
You can type make checkconfig to check your asterisk
configuration now.
-bk
Steve Totaro wrote:
The README is here: svn co
http://svn.digium.com/svn/asterisk-gui/branches/asterisknow
/Configuration
=
You may install sample
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