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) and have a separate
'event' driven log for call logging? Or do
Hi,
On my system I've got :
CLI dahdi show status
Description Alarms IRQbpviol CRC4 Fra
Codi Options LBO
B4XXP (PCI) Card 0 Span 1OK 0 0 0 CCS
AMI YEL 0 db (CSU)/0-133 feet (DSX-1)
B4XXP (PCI) Card 0 Span 2
Hi
Is there a user client that a group, like customer service can use?
We have today an avaya IP-office with phonemanager pro and I want something
equal to phonemanager pro, where you can logon to ques and see how many calls
is in that queue and so on.
Regards
/ralf
Hello David,
Welcome to the world of *. :)
a great day,
v
On Tue, Dec 2, 2008 at 10:40 PM, David fire [EMAIL PROTECTED] wrote:
hi
this is mi first email and just for say hello.
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world
Hello,
I need help for that error message:
chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE
to
My network is:
Client1--
---asterisk1--asterisk2
Client2--
· With client1, I do a call
· Asterisk1 forward the call to
Billing and logging should not be confused theoretically - I agree. But
in practice,
the logging of the calls (not other events of the system) IS used for
billing purposes.
The start and finish time is not enough for many (I not that it is not
enough for me).
The accountcode is not enough for
We have recorded wav files with 44k, 22k, 16k, 11k and 8k
Asterisk does not accept these wav files. I used sox input.wav
output.gsm to get them to work.
However, the only the 8k file did convert and the quality is poor. How
can I improve the quality?
bye
Ronald
On Wed, 3 Dec 2008, Ralf Träskman wrote:
Hi
Is there a user client that a group, like customer service can use?
We have today an avaya IP-office with phonemanager pro and I want
something equal to phonemanager pro, where you can logon to ques and see
how many calls is in that queue and so
I tried sending faxes through Zoiper (Zoiper to Zoiper) last week
and the program crashed. After an update it stopped crashing but
still could not send a FAX. I then tried Kapanga (the free version has a
limited 30
days FAX sending capability) and it worked. This might be of little use
to you
2008/12/3 [EMAIL PROTECTED] [EMAIL PROTECTED]
I tried sending faxes through Zoiper (Zoiper to Zoiper) last week
and the program crashed. After an update it stopped crashing but
still could not send a FAX. I then tried Kapanga (the free version has a
limited 30
days FAX sending capability)
Hello,
My network is:
Client_SS7_1--
---asterisk1--asterisk2
Client_SS7_2--
· I receive a fax from Client_SS7_1
· Asterisk1 forward the call to asterisk2
· Asterisk2 forward the call to asterisk1
· Then, asterisk2 forward the fax to
Hi,
I would be interested in any reports of anyone getting a T.38 FAX to
send or receive successfully with Zoiper. I've tried to test my T.38
implementation against more than one revision of Zoiper, and I yet to
see it behave sanely.
Steve
Olivier wrote:
2008/12/3 [EMAIL PROTECTED]
the best choice is queuemetrics (http://www.queuemetrics.com/)
you have all the info.
David
2008/12/3 Gordon Henderson
[EMAIL PROTECTED][EMAIL PROTECTED]
On Wed, 3 Dec 2008, Ralf Träskman wrote:
Hi
Is there a user client that a group, like customer service can use?
We have today an
On Wednesday 03 December 2008 04:04:10 Ronald Wiplinger (Lists) wrote:
We have recorded wav files with 44k, 22k, 16k, 11k and 8k
Asterisk does not accept these wav files. I used sox input.wav
output.gsm to get them to work.
However, the only the 8k file did convert and the quality is poor.
From: Doug [EMAIL PROTECTED]
Net Neutrality is great in principle. But ISP's need to
somehow control those few percentage of users who suck down
a huge majority of the bandwidth. It's dollars and cents.
There is a rational solution for the traffic management issue. It just needs
to be
Hi all,
I have browsed through a couple of posts that deal with the failure of
applications that originally worked on asterisk 1.2 but fail on asterisk
1.4, but can't seem to understand what I need to change in my installation.
I also went through the CHANGES.txt file in my asterisk source
Ira wrote:
At 12:44 PM 12/2/2008, you wrote:
At 04:03 12/2/2008, Benny Amorsen wrote:
Doug [EMAIL PROTECTED] writes:
Net Neutrality is great in principle. But ISP's need to
somehow control those few percentage of users who suck down
a huge majority of the bandwidth. It's
Thanks Grey and Philipp
Parsing the channel name is what we have been doing, but this has an
unfortunate dependence on username as opposed to peer name. The username
property of a SIP peer is not very well documented, and when using realtime
SIP, it's an immutable field once loaded into cache.
2008/12/3 Steve Underwood [EMAIL PROTECTED]
Hi,
I would be interested in any reports of anyone getting a T.38 FAX to
send or receive successfully with Zoiper. I've tried to test my T.38
implementation against more than one revision of Zoiper, and I yet to
see it behave sanely.
I could
2008/12/3 Olivier [EMAIL PROTECTED]
2008/12/3 Steve Underwood [EMAIL PROTECTED]
Hi,
I would be interested in any reports of anyone getting a T.38 FAX to
send or receive successfully with Zoiper. I've tried to test my T.38
implementation against more than one revision of Zoiper, and I yet
On Dec 2, 2008, at 6:55 PM, Erik (Caneris) wrote:
Erik -
Have you found RealSpeak to be worth the cost?
Actually my last note was probably a bit misleading because in the
particular cases I mentioned RealSpeak, the platform wasn't Asterisk
and Cepstral wasn't even on the radar.
Hi,
Been playing with Call parking, and I can`t help but wonder if I am doing
something incorrectly. The way I understand it (using default config in
features.conf), is I would transfer a call to extension 700, which would
park the call, tell me 701. I could then hang up, go fetch the fright
Mosiuoa Tsietsi wrote:
I have browsed through a couple of posts that deal with the failure of
applications that originally worked on asterisk 1.2 but fail on asterisk
1.4, but can't seem to understand what I need to change in my
installation. I also went through the CHANGES.txt file in my
The way I made this work was to set up 200 as my parker and I do transfer,
200, transfer.
exten = 200,1,Answer
exten = 200,n,Park(701)
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Wednesday, December 03, 2008 10:33 AM
To: 'Asterisk Users Mailing List -
On Wed, Dec 3, 2008 at 1:47 AM, Sebastian [EMAIL PROTECTED] wrote:
I found other solution, I can use cannel local to dial to an extension with
m parameter, then I can put Ringing as the first thing to do that will
follow processing the next lines of the dialplan, with the m option MOH will
On Wed, Dec 03, 2008 at 11:33:08AM -0500, Mike wrote:
Hi,
Been playing with Call parking, and I can`t help but wonder if I am doing
something incorrectly. The way I understand it (using default config in
features.conf), is I would transfer a call to extension 700, which would
park the
On Wed, Dec 03, 2008 at 10:56:48AM -0600, Danny Nicholas wrote:
The way I made this work was to set up 200 as my parker and I do transfer,
200, transfer.
exten = 200,1,Answer
exten = 200,n,Park(701)
That will work but only for one call park slot. If that's what you
want then great.
If you
I agree with [EMAIL PROTECTED] we need the events to create the final CDR.
I will not waste list space on a long but just show you 2 reallife
examples that can't be handled both within the same 'fixed' way of
generating CDR's as we do now.The new system that 's proposed would
handle both just
This actually works for multiple slots. When 701 is occupied, * finds next
defined slow.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Lister
Sent: Wednesday, December 03, 2008 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
I need to test canreinvite=yes with 2softphones and 1 asterisk.
I want to have that :
http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb
ridge.png
But I have that http://www.zimagez.com/zimage/canreinvite.php
Canreinvite=yes work for all phones or
Hello,
I need to test canreinvite=yes with 2softphones and 1 asterisk..
I want to have that :
http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb
ridge.png
But I have that http://www.zimagez.com/zimage/canreinvite.php
Canreinvite=yes work for all phones or
://lists.digium.com/mailman/listinfo/asterisk-users
__ Information from ESET Smart Security, version of virus signature
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http://www.eset.com
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Mike wrote:
Can`t the parked call just go park itself (and hang up my leg of the
call), and ideally call me back if not picked up within x seconds?
Look at the parkcall option under the features.conf
parkcall = ## ; Park call (one step parking)
Doug
--
Ben
Someone have a solution for me ?
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de BERGANZ
François
Envoyé : mercredi 3 décembre 2008 18:24
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] canreinvite=yes problem
Hello,
I need to test canreinvite=yes with
On Wed, Dec 3, 2008 at 7:27 PM, Sebastian [EMAIL PROTECTED] wrote:
The thing is I have to wait checking a database value to change the state,
that duration is not long, but on any case I don't know when will be ready
to go on.
If I use MusicOnHold app the dialplan get stuck there and there's
On Tue, Dec 02, 2008 at 05:04:25PM +0530, Max Alex wrote:
Hi All,
I need to stop the transfer feature on particular sip user.
I am using linksys phone and it has set the forwarding enable to another
user.
I have three users 2101, 2102, 2103.
2102 is registered in linksys phone with
On Wed, Dec 03, 2008 at 11:13:49AM -0600, Danny Nicholas wrote:
This actually works for multiple slots. When 701 is occupied, * finds next
defined slow.
Does it announce what that slot is before doing it?
Rob
--
Robert Lister - London Internet Exchange - http://www.linx.net/
sip:[EMAIL
Hello,
canreinvite, don't work with all softphone or hardphone.
Regards
On Wed, Dec 3, 2008 at 12:38 PM, BERGANZ François
[EMAIL PROTECTED] wrote:
Someone have a solution for me ?
*De :* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *De la part de* BERGANZ François
*Envoyé :* mercredi
Yes it does. It plays the slot number, then does music on hold until you
hit transfer again.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Lister
Sent: Wednesday, December 03, 2008 12:03 PM
To: Asterisk Users Mailing List - Non-Commercial
On Wed, Dec 03, 2008 at 06:23:32PM +0100, BERGANZ François wrote:
Hello,
I need to test canreinvite=yes with 2softphones and 1 asterisk.
I want to have that :
http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outbridge.png
But I have that
OK, I can park the calls OK, but I don't get the announcement -- I am
using freepbx if that makes any difference.
on Wednesday 12/03/2008 Doug Lytle([EMAIL PROTECTED]) wrote
Mike wrote:
Can`t the parked call just go park itself (and hang up my leg of the
call), and ideally
On Wed, Dec 03, 2008 at 03:27:28PM -0200, Sebastian wrote:
The thing is I have to wait checking a database value to change the state,
that duration is not long, but on any case I don't know when will be ready
to go on.
If I use MusicOnHold app the dialplan get stuck there and there's no
canreinvite=yes should work as long as 1) there is no NAT involved
anywhere in the call path, 2) All legs of the call are using the same
codec, 3) you do not have the t/T/w/W (and maybe a few other) options to
the Dial line.
Remember the only way you can really tell if a reinvite happens is by
John covici wrote:
OK, I can park the calls OK, but I don't get the announcement -- I am
using freepbx if that makes any difference.
If you park a call and do not hear the announcement then you are doing a
BLIND transfer, not an ATTENDED transfer. You should be doing attended
transfers for
John covici wrote:
OK, I can park the calls OK, but I don't get the announcement -- I am
using freepbx if that makes any difference.
Apparently it does. What does it show on the console when doing the one
step parking? Also make sure your dial command has kK:
k- Allow the called
hi
sorry about the urgent but it is urgent
i have problems configuring a connection between asterisk and avaya using
H323.
the module i am usign is ooh323
what do you need to help me?
and any tip or hint?
thanks!!!
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
Does anyone have the SIP firmware for a Mitel 5340?
Thanks,
Craig Van Ham
Network Operations
PH 1-306-931-8822 Ext: 14
Toll Free: 1-866-328-6144 Ext:14
Email: [EMAIL PROTECTED]
Note: The information contained in this e-mail is confidential and may
be subject to the rules of
On Sun, Nov 16, 2008 at 8:55 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Sun, Nov 16, 2008 at 4:28 AM, Sriram [EMAIL PROTECTED] wrote:
Hi
below are my configs:
pstn(e1)---asterisk (span1)-legacy pbx(connected via span2)-
legacy pbx analog extensions.
my dial plan is like
Hi,
How do I disabled asterisk to use database and storage voicemail in
directory.
Im getting the below error
[Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to
obtain database object for 'asterisk'!
[Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353
The Asterisk.org development team has released Asterisk version 1.6.0.3-rc1.
This release is available for immediate download from
http://downloads.digium.com/.
This release candidate follows on the recent (broken) release of 1.6.0.2 with
multiple fixes. This release also marks the first time
I would go for chan_h323. Much more stable since 1.4
and the config more close to the other channel configs too.
We used it on production for a long time and it worked well
although a little heavy cpu-wise. To get started you need to install
openh323
and pwlib from here
thanks you very much i will try it tomorrow when i reach the office.
David
2008/12/3 [EMAIL PROTECTED] [EMAIL PROTECTED]
I would go for chan_h323. Much more stable since 1.4
and the config more close to the other channel configs too.
We used it on production for a long time and it worked
- [EMAIL PROTECTED] escribió:
I would go for chan_h323. Much more stable since 1.4
and the config more close to the other channel configs too.
We used it on production for a long time and it worked well
although a little heavy cpu-wise. To get started you need to install
openh323
and
gracias!!!
(thanks)
2008/12/3 Guillermo V. Salas [EMAIL PROTECTED]
- [EMAIL PROTECTED] escribió:
I would go for chan_h323. Much more stable since 1.4
and the config more close to the other channel configs too.
We used it on production for a long time and it worked well
although a
Ironically, much of the disagreement comes from everyone being *right*.
Seriously.
Philisophically, Asterisk is a tool chest, not a true product. As an
analogy, if one wants to sit, one can either buy a chair or get a
saw/hammer/nails/lumber and build one. Both will do the job. Asterisk is
my dial plan is like callers dial into asterisk(span1) , hear an IVR
option and they are connected to the agents via the legacy pbx (which is in
sync with asterisk on span2)This works perfectly fine until about 200
calls or so...After that time when asterisk tries to dial to the legacy pbx
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__ Information from ESET Smart Security, version of virus signature
database 3660 (20081203) __
The message was checked by ESET Smart Security.
http://www.eset.com
__ Information from ESET
By legacy phone I assume you have an analog card connected to your
Asterisk server. I've not used analog phones with Asterisk in many
years, but IIRC you need transfer=yes and threewaycalling=yes in
zapata.conf/chan_dhadi.conf. You would then do a 2nd flash to complete
the transfer. On
Yep, those are fine and as I say, it does actually park the call
because I can hang up and type 701 and get the call back, but my only
problem is it hangs up immediately instead of playing the
announcement.
on Wednesday 12/03/2008 Eric \ManxPower\ Wieling([EMAIL PROTECTED]) wrote
By legacy
visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__ Information from ESET Smart Security, version of virus signature
database 3660 (20081203) __
The message was checked by ESET Smart Security.
http://www.eset.com
__ Information from ESET Smart
Installing 1.4.23-rc2, I actually looked at the startup and saw this
warning:
WARNING[10730]: loader.c:359 load_dynamic_module: Error loading module
'app_directory.so': /usr/lib/libc-client.so.2007: undefined symbol: mm_dlog
I'm running Fedora Core 9, with libc-client 2007d. googling didn't
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I'm have a bit of a problem with temporary greetings.
I'm using 1.6.0.1 with IMAP storage. If I go into comedian mail and
record a temporary greeting, I get a Message [0] on my IMAP server. In
addition, I get
:
http://lists.digium.com/mailman/listinfo/asterisk-users
__ Information from ESET Smart Security, version of virus
signature
database 3660 (20081203) __
The message was checked by ESET Smart Security.
http://www.eset.com
__ Information from ESET Smart Security
What sip client are you using on WM6 side ?
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Security, version of virus
signature
database 3660 (20081203) __
The message was checked by ESET Smart Security.
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signature
database 3662 (20081203) __
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I'm using the Wm6 built in client. (Enabled via CAB file to add-back files
removed from ROM)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hakem Ta
Sent: December 3, 2008 8:42 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote host
Microsoft doesn't make a native SIP client in Windows Mobile you can use
for a phone call. Do you mean Windows Live Messenger?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG
Technical Support
Sent: Wednesday, December 03, 2008 9:15 PM
To:
You'll have to recheck your facts...MS does include a SIP client in WM6.
And it works great J Carriers/brands can remove items from ROM, but the SIP
client is in by default.
Have a look on XDA developers web site for details
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi I have problem with TE121 Digium card. I connected it to modem keymile
Music 200 (provided by telco) but I can see 2 red lights on modem (both
bellow words rx) and my card is red too. I tried to make experiment and made
loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope
69 matches
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