Happy birthday asterisk!
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de SIP
Envoyé : vendredi 5 décembre 2008 06:14
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Friday, Asterisk is 9 years old!
randulo
Hi
Hm is this function for recording? The thing I want to do is to be able to see
how many calls there is waiting in a queue, maybe im looking in the wrong
direction?
/ralf
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Kauffmann
Sent: den
I think you may have misunderstood me. I didn't say don't have the
extra information, I said Let's have the 'extra' information in a
different way and leave the existing CDR's as they are.
Take the example of a 'real' PBX - the SDX/Lucent/Avaya Index. The
Index had 2 options for 'logging' -
You are looking in the wrong place.
Have a look at the following:
Core show function QUEUE_WAITING_COUNT
-= Info about function 'QUEUE_WAITING_COUNT' =-
[Syntax]
QUEUE_WAITING_COUNT(queuename)
[Synopsis]
Count number of calls currently waiting in a queue
[Description]
Returns the number
Hi,
I am developing asterisk support for our application using the Async AGI
and Asterisk-Java.
One thing I haven't been able to implement is how to stop playing a
sound. Something similar to StopIO for Dialogic GlobalCall or
DivaStopSending for Eicon.
Is there any way to achieve this today which
Ralf Träskman wrote:
Hi
Hm is this function for recording? The thing I want to do is to be able to
see how many calls there is waiting in a queue, maybe im looking in the wrong
direction?
/ralf
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
hi,
i have a problem, and i am completely stuck with it, i hope someone can
point out where is my config wrong.
I have three server, connect together with IAX trunking. The server are
at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and serbia
(10.0.3.4, V1.4.22). I have a hardphone
On Fri, Dec 5, 2008 at 8:26 AM, Andrew Thomas [EMAIL PROTECTED] wrote:
In summary: Leave CDR exactly as it is and create a new CEL (Call Event
Logging) module (optional in modules.conf) that puts out (and does not
accept) call event information (ie. a one-way fire-and-forget output
from
Hi all,
I'm testing Linksys SPA922 phone and I have strange issue. when call is
finished on the phone I see CallEnded and normal silence for cca. 5 seconds
and then I get fast busy for cca. 20 sec. So, this isn't automatic hangup as on
other phones I have tried (Cisco 7940, grandstream,
Hello,
Asterisk 1.4.18.1
PWlib 1.10.0
Openh323 1.18.0
../asterisk/channels/h323 compiled from source.
Under high load H323 crashes and kills Asterisk, debug shows:
(gdb) bt
#0 0x007a2b18 in strcmp () from /lib/libc.so.6
#1 0x014478a1 in find_call_locked (call_reference=13,
Is it possible to check certain varibles on the live system, for example, what
the current setting for pridialplan is? I know what is set in the config files,
but the behaviour does not reflect this. Can this be checked?
Kind Regards:
Gabriel
Thanks for this Greyman - it's all beginning to make sense now ;).
I agree that the 'loss of CDR upon txfr' is a nasty bug which does need
to be addressed before anything else (assuming it hasn't been already).
But, wouldn't it be better if you could ignore the CDR's completely and
use an event
Lincoln King-Cliby wrote:
Hi All,
I know this is possible when picking up a parked call, but I haven’t
found any information re: transferred calls.
Our operator ususally says to the recipient, Okay, here he/she is and
hits the transfer. Nobody has complained yet.
But, to answer your
Right after a bit of investigation i've found that it's because we're
running a mysql database on the same server, it was fine all morning with a
relatively low load on the server, now the rest of the agents have logged in
the problem has returned!
Time to buy a new database server... mystery
Hello Andy,
But, wouldn't it be better if you could ignore the CDR's completely and
use an event based system? This would give you ALL the information you
need. All that remains is to filter out the un-required bits.
I'd disagree. In some cases a event based system would be the best
Hello,
I think, that I find a bug for a specific asterisk use.
My network test is:
Client 1
--asterisk0(client)---asterisk1(ss7)--Asterisk2(media)
Client 2
I need that a call from Client1 go through
asterisk0-asterisk1-asterisk2-asterisk1-asterisk0-Client2
When I
Quote : Like I said earlier - the CDR's aren't
reliable enough for a billing platform (as you've
rightly pointed out) but are OK for very basic call
logging (something the customer can look at).
I couldn't disagree more. The CDRs is the MOST reliable
source for billing purposes (in postpaid
Hello List,
we are in the need to reach an external Conference-System, whos
numbering system is *2*. Unfortunately *2 is the featurecode for
attended transfer in our local asterisk, so the call doesn't come
through. Is there a way to somehow escape the featurecode, so we can
reach the
Dear All,
We are trying to install CDR Stats module. We are able to open its web pages
but unable to retrieve data from CDRs. Can anybody suggest that how to
connect this module with Asterisk CDRs database ?
Regards,
--
Kashif Naeem
Business Development Manager
Hadi Telecom
Olivier wrote:
2008/12/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
2008/12/3 Steve Underwood [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
Hi,
I would be interested in any reports of anyone getting a T.38
FAX to
send or receive successfully
here you go.
http://www.areski.net/asterisk-stat-v2/about.php
On Fri, Dec 5, 2008 at 5:08 PM, Kashif Naeem [EMAIL PROTECTED] wrote:
Dear All,
We are trying to install CDR Stats module. We are able to open its web
pages but unable to retrieve data from CDRs. Can anybody suggest that how
2008/12/5 dubravko caric [EMAIL PROTECTED]:
Hi all,
I'm testing Linksys SPA922 phone and I have strange issue. when call is
finished on the phone I see CallEnded and normal silence for cca. 5
seconds and then I get fast busy for cca. 20 sec. So, this isn't automatic
hangup as on other phones
Quote : I couldn't disagree more. The CDRs is the MOST reliable
source for billing purposes
...at the moment. Have you read about Greyman's transfer problem?
If you are billing customers purely on the CDR output from Asterisk -
then good luck to you :).
I'd disagree. In some cases a event based system would be the best
solution, but in systems with high call volumes, scanning through events
looking for the proper billing information and parsing them would be a
hard job compared to CDRs.
That's just it - you wouldn't be 'scanning' any CDR's -
On Fri, Dec 5, 2008 at 2:35 PM, Andrew Thomas [EMAIL PROTECTED] wrote:
I'd disagree. In some cases a event based system would be the best
solution, but in systems with high call volumes, scanning through events
looking for the proper billing information and parsing them would be a
hard job
Hello Henrik,
I have not used Asterisk from a user perspective lately, but, when I
added the async agi functionality, I used to control this using a
manager redirect action to the same priority where the channel calls
async agi, that will work like a break that re-enters the async agi
loop .
Thanks for reply. I think I already made a crossover cable :
pins 1,2,4,5 and 4,5,1,2 and I still see red light on modem.
pin 1 is in rj45-1 and is going to pin 4 in rj45-2 (orange with white
stripe)
pin 2 is in rj45-1 and is going to pin 5 in rj45-2 (solid orange)
pin 4 is in rj45-1 and is going
hi
the line We think we are cpe but they think they are cpe too
means in the mmm /etc/asterisk/chan_dahdi.conf or in
/etc/asterisk/zapata.conf you told the the card act as cpe but in a
conection mst be one cpe and one net if both are net or you make a loopback
you will have that.
and the colores
Top posting strikes again:
On Fri, Dec 05, 2008 at 01:39:59PM +0200, [EMAIL PROTECTED] wrote:
Quote : Like I said earlier - the CDR's aren't
reliable enough for a billing platform (as you've
rightly pointed out) but are OK for very basic call
logging (something the customer can look at).
On Fri, Dec 5, 2008 at 8:14 AM, Uros Djokic [EMAIL PROTECTED] wrote:
Thanks for reply. I think I already made a crossover cable :
pins 1,2,4,5 and 4,5,1,2 and I still see red light on modem.
pin 1 is in rj45-1 and is going to pin 4 in rj45-2 (orange with white
stripe)
pin 2 is in rj45-1 and
But the new CDRs that we are discussing would have to deal
with transfers correctly. I think that's where the whole thing started.
I am not happy with the current CDRs system either. I find it obsolete.
That is why I am not using it for billing purposes. But a NEW one that
meets certain criteria
Pardon me,
Granted ;).
I have created realtime stats package that's based on CDR, you see new
info immediately after call leg/event is over
I see what you are saying but can you show hold-times etc? For example,
call comes in to A, A puts call on hold, A dials B, B answers A, A
transfers call
Quote : That's just it - you wouldn't be 'scanning' any CDR's - you'd be given
Events. Your 3rd party app could then do anything it wanted to with
them.
A 3rd party live application introduces one more
point of failure in the whole process. A 3rd party CDRs
aggregator can run at its own pace
Tzafrir Cohen wrote:
Top posting strikes again:
On Fri, Dec 05, 2008 at 01:39:59PM +0200, [EMAIL PROTECTED] wrote:
Quote : Like I said earlier - the CDR's aren't
reliable enough for a billing platform (as you've
rightly pointed out) but are OK for very basic call
logging (something the
Address added to spam filter. Please do NOT e-mail me again.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 05 December 2008 13:27
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] top posting again [was: Re: CDR Design]
Amen!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Apostolos
Pantsiopoulos
Sent: 05 December 2008 13:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR Design
Quote : That's just it - you
- Original Message -
From: Andrew Thomas [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 5 December, 2008 13:49:59 GMT +00:00 GMT Britain, Ireland,
Portugal
Subject: Re: [asterisk-users] top posting again
- Original Message -
From: Mr Gabriel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 5 December, 2008 13:58:09 GMT +00:00 GMT Britain, Ireland,
Portugal
Subject: Re: [asterisk-users] top posting again
I see a variety of DECT 6 phones available CHEAP at costco. Is there a way
to convert these to SIP?
I recall someone talking about a Siemens devices that works with all DECT
phones, making them SIP (I think)
___
-- Bandwidth and
Carsten Maass wrote:
we are in the need to reach an external Conference-System, whos
numbering system is *2*. Unfortunately *2 is the featurecode for
attended transfer in our local asterisk, so the call doesn't come
through. Is there a way to somehow escape the featurecode, so we can
You probably cant' convert then easily. However, several comapnies
offer SIP/DECT phones...not as cheap as Costco, but more capable. I
have a snom m3 system and a Siemens S685IP.
You can get the costco phone and add an ATA, turning them into SIP
capable. Adds the cost of the ATA.
The Siemens
Hold down 2,4,6,8 and * at the same time. This is the 501 reset key
sequence.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, December 04, 2008 6:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
On Fri, Dec 5, 2008 at 3:41 PM, Andrew Thomas [EMAIL PROTECTED] wrote:
Pardon me,
Granted ;).
I have created realtime stats package that's based on CDR, you see new
info immediately after call leg/event is over
I see what you are saying but can you show hold-times etc? For example,
call
Thanks for that, it IS appreciated - but, everyone, can we please not argue
this matter any more. Some see it as top posting - some don't. I really don't
care either way.
No if we could just get back to the subject in hand and not clog up this list
with flames.
Thanks
Andy
-Original
Have a look at ATA devices. Any good VoIP equipment reseller should have them
available.
http://www.voip-info.org/wiki-ATA is worth a look.
Cheers
Andy
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: 05 December 2008 14:17
Friday, December 5, 2008, 2:49:59 PM, Andrew wrote:
Address added to spam filter. Please do NOT e-mail me again.
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
A: Top-posting.
Q: What is the most annoying thing in e-mail?
--
Best
Q: What is the most annoying thing in e-mail?
Spam and useless replies when I've already asked for this topic to be
closed *sigh*.
-- -Original Message-
-- From: [EMAIL PROTECTED]
[mailto:asterisk-users-
-- [EMAIL PROTECTED] On Behalf Of Gergo Csibra
-- Sent: 05 December 2008 14:41
On Fri, Dec 5, 2008 at 3:47 PM, Apostolos Pantsiopoulos [EMAIL PROTECTED]
wrote:
Tzafrir Cohen wrote:
Top posting strikes again:
On Fri, Dec 05, 2008 at 01:39:59PM +0200, [EMAIL PROTECTED] wrote:
Quote : Like I said earlier - the CDR's aren't
reliable enough for a billing platform (as
Hi Moy,
Thank you for your quick answer. Also thanks for implementing the great
async agi functionality!
Yes, this works good for me. A StopIO feature would of course be
cleaner but this certainly does the trick.
Regards,
Henrik
Moises Silva skrev:
Hello Henrik,
I have not used
H, not sure about you but I often pick up a book and flick from
the back to the front, does nobody else do that?
On 05/12/2008, Atis Lezdins [EMAIL PROTECTED] wrote:
On Fri, Dec 5, 2008 at 3:47 PM, Apostolos Pantsiopoulos [EMAIL PROTECTED]
wrote:
Tzafrir Cohen wrote:
Top posting
On Dec 4, 2008, at 7:31 PM, Matt Gibson wrote:
We often find ourselves reading through all sorts of contests on the
Internet that never seem to echo our own personal skill set or
interests.
Perhaps you've even fantasized about a type of contest with the
types of
prizes and goodies that
Hello,
Andrew Thomas wrote:
I'd disagree. In some cases a event based system would be the best
solution, but in systems with high call volumes, scanning through events
looking for the proper billing information and parsing them would be a
hard job compared to CDRs.
That's just it -
Atis Lezdins wrote:
When i started to
write this implementation, luckily i didn't had much expertise in
telephony, so i did it from programmers point of view. There's even
funny story about this in our company - we had some Project managers
and Development managers hired later who had lots
Is there a way, for debugging purpose, to have a level where only Noop()
cmds are shown in the CLI but nothing else in the dialplan appears (except
for errors and warnings or course)?
Mike
___
-- Bandwidth and Colocation Provided by
Hello!
I have a problem with build astersik-addons-1.4.7 on Solaris 10. When I
tried to do make I got such error:
*
chan_ooh323.c: In function `reload_config':
chan_ooh323.c:2053: error: `IPTOS_MINCOST' undeclared (first use in this
function)
chan_ooh323.c:2053: error: (Each undeclared
Andrew Thomas wrote:
Quote : I couldn't disagree more. The CDRs is the MOST reliable
source for billing purposes
...at the moment. Have you read about Greyman's transfer problem?
If you are billing customers purely on the CDR output from Asterisk -
then good luck to you :).
This is
Anthony Francis wrote:
Atis Lezdins wrote:
When i started to
write this implementation, luckily i didn't had much expertise in
telephony, so i did it from programmers point of view. There's even
funny story about this in our company - we had some Project managers
and Development managers
2008/12/5 Stefan Lekov [EMAIL PROTECTED]
Olivier wrote:
2008/12/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
2008/12/3 Steve Underwood [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
Hi,
I would be interested in any reports of anyone getting a T.38
FAX
On Fri, 05 Dec 2008 10:59:54 -0500, Neil Fusillo wrote:
Michael,
Was there something particularly special you had to do to get your M3 to
work? I'm now on my second one from E4 Technologies (from whom I'm still
waiting for a return service call), and both of them, after following
the
Another thing to be aware of as the wish list for the Asterisk CDR
continues to grow is that right now Asterisk does not lend itself to
accurately creating the most fundamental requirement of a CDR which is
to accurately record at the very least the originator, destination,
time and duration for
Sometimes I do. It depends on my mood and purpose. And sometimes the
author prefers to write last things first, for whatever reason.
I'm kind of agnostic, too.
Mike Dent wrote:
H, not sure about you but I often pick up a book and flick from
the back to the front, does nobody else do that?
Henrik Westerberg wrote:
Yes, this works good for me. A StopIO feature would of course be cleaner
but this certainly does the trick.
The ExternalIVR interface, while not quite as feature-filled as AGI,
does in fact work in a true non-blocking fashion, and supports exactly
what you are looking
Or just type slowly enough -- I think the timeout is half a second or
so.
on Friday 12/05/2008 Matthew J. Roth([EMAIL PROTECTED]) wrote
Carsten Maass wrote:
we are in the need to reach an external Conference-System, whos
numbering system is *2*. Unfortunately *2 is the featurecode
On Fri, 05 Dec 2008 11:37:26 -0500, SIP wrote:
Firmware's set to 1.11.
Of course, there's no way to update it, since I can't get the networking
to work properly.
I'd put it on LAN using static IP, then set the base to get the latest
firmware.
Michael
--
Michael Graves
mgravesatmstvp.com
On Dec 5, 2008, at 11:31 AM, Michael Graves wrote:
On Fri, 05 Dec 2008 10:59:54 -0500, Neil Fusillo wrote:
Michael,
Was there something particularly special you had to do to get your
M3 to
work? I'm now on my second one from E4 Technologies (from whom I'm
still
waiting for a return
Is there any way to provide the user receiving an attended transfer
with a tone or other audible indication that the transfer is
completed (i.e. Party A calls Party B, Party B announces the call
while transferring to Party C, Party C hears tone when Party B
completes the transfer so
Fred Posner wrote:
On Dec 5, 2008, at 11:31 AM, Michael Graves wrote:
On Fri, 05 Dec 2008 10:59:54 -0500, Neil Fusillo wrote:
Michael,
Was there something particularly special you had to do to get your M3 to
work? I'm now on my second one from E4 Technologies (from whom I'm still
waiting
I will publish a tutorial in the beginning of next week about how to
configure Zoiper and Asterisk to do t.38 together. (and while doing so
test the latest version again to make sure it really works)
Feel free to send us any bugtickets if you think something is broken, in
the case of t.38
Thanks for the answer Terry, it's kind of what I expected. I may have to look
into using Attended transfers in Asterisk, but I think my users really prefer
having the TRNSFR soft key instead of remembering a feature code.
I guess then the next question... Does anyone know of a way to map the
I guess there is a variety of opinions on this, some of which relates to the
tools a person is using. The absolutely most offensive thing to me in a post
is to have to scroll through a bunch of copied original material that I've
already read six times to get to the new part. My own preference
Grey Man wrote:
Another thing to be aware of as the wish list for the Asterisk CDR
continues to grow is that right now Asterisk does not lend itself to
accurately creating the most fundamental requirement of a CDR which is
to accurately record at the very least the originator, destination,
Hi,
I've noticed that if I have a multi-line linksys (942 or 962) phone
with the same sip registration mapped to each line key, that if all
the lines are full the phone will accept another call. I would expect
the phone to respond with busy so the call would to directly to
voicemail.
Has
2008/12/5 [EMAIL PROTECTED] [EMAIL PROTECTED]
I will publish a tutorial in the beginning of next week about how to
configure Zoiper and Asterisk to do t.38 together. (and while doing so
test the latest version again to make sure it really works)
Fine !
I'll wait for it and report
hi
in sip.conf there is a parameter calllimit or something like that use it...
David
2008/12/5 James Lamanna [EMAIL PROTECTED]
Hi,
I've noticed that if I have a multi-line linksys (942 or 962) phone
with the same sip registration mapped to each line key, that if all
the lines are full the
On Friday 05 December 2008 11:11:33 Wilton Helm wrote:
I guess there is a variety of opinions on this, some of which relates to
the tools a person is using. The absolutely most offensive thing to me in
a post is to have to scroll through a bunch of copied original material
that I've already
Tilghman wrote
...will you ask your question in accordance with the rules that were set
out in advance...?
Where can we review these rules?
--Don
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
i have the solution so every one is happy i will write over and below :- )
2008/12/5 Tilghman Lesher [EMAIL PROTECTED]
On Friday 05 December 2008 11:11:33 Wilton Helm wrote:
I guess there is a variety of opinions on this, some of which relates to
the tools a person is using. The
On Fri, Dec 05, 2008 at 10:11:33AM -0700, Wilton Helm wrote:
I guess there is a variety of opinions on this, some of which relates
to the tools a person is using. The absolutely most offensive thing
to me in a post is to have to scroll through a bunch of copied
original material that I've
Have you checked voip.org? They have this kind of information for a
Polycom, so they probably have similar information for the Cisco 79x1.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lincoln
King-Cliby
Sent: Friday, December 05, 2008 11:10 AM
To:
selected as
the winners and will be awarded the following prizes:
[snip]
I think you'd get just as much interest in an Obfuscated Dialplan
Contest which seems to be the most popular type of dialplan
programming. The more unreadable, ugly, and opaque the code becomes,
the more
Good programmers can diagram the most obfuscated code. It's part of the
job description.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: Friday, December 05, 2008 12:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
On Friday 05 December 2008 11:57:56 Don Kelly wrote:
Tilghman wrote
...will you ask your question in accordance with the rules that were set
out in advance...?
Where can we review these rules?
This one is as good as any:
http://db.tidbits.com/article/5386
Note the publication date in
On Fri, Dec 05, 2008 at 12:46:26PM -0600, Danny Nicholas wrote:
Good programmers can diagram the most obfuscated code. It's part of the
job description.
Anybody with a dialplan that looks like a puppy?
Reminder from a previous thread: a really silly script to graph (using
gnuplot) inclusions
I have played with both call-limit=2 and busylevel=1 and have yet been able
to get a busy returned from a call attempt. If someone actually has this
working some insight into how you are setup would be much appreciated. I can
cause a channel unavailable but not a busy.
--
Jim Dickenson
David fire [EMAIL PROTECTED]
hi
in sip.conf there is a parameter calllimit or something like that use it...
I believe the SIP call-limit parameter drops the call if the call
limit is exceeded and does not respond as if the phone were busy.
Also, since I have different models of phones with
On Fri, Dec 5, 2008 at 4:38 PM, John Todd [EMAIL PROTECTED] wrote:
I think you'd get just as much interest in an Obfuscated Dialplan
Contest which seems to be the most popular type of dialplan
programming. The more unreadable, ugly, and opaque the code becomes,
snip
I prefer the well
On Fri, Dec 5, 2008 at 8:12 PM, randulo [EMAIL PROTECTED] wrote:
On Fri, Dec 5, 2008 at 4:38 PM, John Todd [EMAIL PROTECTED] wrote:
I think you'd get just as much interest in an Obfuscated Dialplan
Contest which seems to be the most popular type of dialplan
programming. The more unreadable,
On Fri, 5 Dec 2008, Tzafrir Cohen wrote:
On Fri, Dec 05, 2008 at 12:46:26PM -0600, Danny Nicholas wrote:
Good programmers can diagram the most obfuscated code. It's part of the
job description.
Anybody with a dialplan that looks like a puppy?
Reminder from a previous thread: a really
El vie, 05-12-2008 a las 19:04 +0300, Mikhail Zhirnov escribió:
make[2]: cc: Command not found
Looks like you need cc installed.
Best regards,
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esquina
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 7815
Celular : +593
I agree with the fact that the base is broken and needs to be fixed
first.
--
We wouldn't have this discussion if we had a close to perfect CDR design
that just needed some 'fixing'.
The processes of just adding another couple of patches has been ongoing
for more than year now.
I think
On Wed, 2008-12-03 at 08:11 +, Andrew Thomas wrote:
It seems to me that we are confusing billing and logging here. Call
billing only really needs the start and finish (like we get now) - but
proper call logging requires all steps.
Do we leave CDR's as they are (for billing purposes)
On Fri, 2008-12-05 at 13:39 +0200, [EMAIL PROTECTED] wrote:
Quote : Like I said earlier - the CDR's aren't
reliable enough for a billing platform (as you've
rightly pointed out) but are OK for very basic call
logging (something the customer can look at).
I couldn't disagree more. The CDRs
On Fri, 2008-12-05 at 10:52 +, Andrew Thomas wrote:
Thanks for this Greyman - it's all beginning to make sense now ;).
I agree that the 'loss of CDR upon txfr' is a nasty bug which does need
to be addressed before anything else (assuming it hasn't been already).
But, wouldn't it be
On Fri, Dec 05, 2008 at 07:24:52PM +, Jeff LaCoursiere wrote:
Many years ago when I was in school we had an obfuscated 'C' contest, and
I recall one year the winning entry had #define'd all of the neccessary
code into morse, so that the eventual .c file was completely in morse
code.
Hi all,
I've just upgraded to latest 1.6.0 SVN from a few days ago and my Gosubs
have stopped working.
This is from the verbose logs:
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/aaisp-3802,
1?5:7) in new stack
-- Goto (incoming-aaisp,0407271,5)
-- Executing [EMAIL PROTECTED]:5]
On Fri, 2008-12-05 at 12:35 +, Andrew Thomas wrote:
I'd disagree. In some cases a event based system would be the best
solution, but in systems with high call volumes, scanning through events
looking for the proper billing information and parsing them would be a
hard job compared to
makes sense
On Fri, Dec 5, 2008 at 7:59 PM, David fire [EMAIL PROTECTED] wrote:
i have the solution so every one is happy i will write over and below :- )
2008/12/5 Tilghman Lesher [EMAIL PROTECTED]
On Friday 05 December 2008 11:11:33 Wilton Helm wrote:
I guess there is a variety of
On Fri, 2008-12-05 at 14:47 +0200, Atis Lezdins wrote:
On Fri, Dec 5, 2008 at 2:35 PM, Andrew Thomas [EMAIL PROTECTED] wrote:
I'd disagree. In some cases a event based system would be the best
solution, but in systems with high call volumes, scanning through events
looking for the proper
Hello!
I have a problem with build astersik-addons-1.4.7 on Solaris 10. When I
tried to do make I got such error:
*
chan_ooh323.c: In function `reload_config':
chan_ooh323.c:2053: error: `IPTOS_MINCOST' undeclared (first use in this
function)
chan_ooh323.c:2053: error: (Each undeclared identifier
On Fri, 2008-12-05 at 15:32 +0200, Apostolos Pantsiopoulos wrote:
But the new CDRs that we are discussing would have to deal
with transfers correctly. I think that's where the whole thing started.
I am not happy with the current CDRs system either. I find it obsolete.
That is why I am not
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