Jean-Michel Hiver wrote:
But then again come to them with a few million monthly minutes under
your belt and I'm sure they'll change the TOS for you...
Maybe not, as the ToS also state:
The customer agrees to purchase VoipJet termination in small amounts
What does this mean? We have to start with
I attempted setting up a queue with agents that log in, and
get called with incoming calls:
Agents log in using:
exten = *88,1,AgentCallbackLogin(${CALLERIDNUM}|[EMAIL PROTECTED])
Calls get into the queue with:
exten = 6029995654,1,Queue(test-noc|t|||60)
queues.conf:
[test-noc]
strategy
On 15 Mar 2005, at 23:52, Giudice, Salvatore wrote:
we were able to handle a peak of 700k inserts per hour. MySQL gave us
very few problems and probably had a cumulative downtime of
approximately 4 days per year until the project was decommissioned.
When
y
That's more than 1% downtime, not even
On Wed, Mar 16, 2005 at 03:25:17PM +0800, Ronald Wiplinger wrote:
Mar 16 15:13:45 DEBUG[29502]: Raw Hangup 69.73.19.178:4569, src=14, dst=1259
Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Update SQL: UPDATE
sip_buddies SET name = '621' WHERE allow = 'g729'
Mar 16 15:13:45 DEBUG[29502]: MySQL
hello
i try to call from sip phone on asteris to open phone
on GnuGK.
can any one tell me why it is saying
chan_oh323.c:2501 ast_oh323_new: Internal channel
initialization failed. Bad binary?
Mar 16 13:28:46 WARNING[5963]: chan_oh323.c:2727
oh323_request: Failed to create new H.323 private
Martijn van Oosterhout wrote:
On Wed, Mar 16, 2005 at 03:25:17PM +0800, Ronald Wiplinger wrote:
Mar 16 15:13:45 DEBUG[29502]: Raw Hangup 69.73.19.178:4569, src=14, dst=1259
Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Update SQL: UPDATE
sip_buddies SET name = '621' WHERE allow = 'g729'
Mar 16
Hi,
I have about one year of experience with Asterisk, working with ZAP
(digium, junghanns) ZAPHFC, SIP and IAX. These technologies are quite
clear to me, the problem is that I have no experience with H323, but
now, I need to use this also.
The problem that I have is very trivial, so I think
Hello
I'm interested in setting up a calling card application on asterisk. I
noticed a number in the wiki, both free and commercial. To experiment with,
I'm after a GNU licenced app...Which one would you recommend ?
Regards..Peter
--
Open WebMail Project (http://openwebmail.org)
Greetings *`s,
There was a thread some time back about making calls via * from a web
interface...ie user clicks number on web page and call is made...
I`ve googled with a few words, checked the wiki, and tried to scan
through the archives, but no joy...
Any links/pointers/keywords
David Zanetti wrote:
I've been beating my head a bit against the 1.0.6 Debian builds of
Asterisk, using an E100P (E1, single span) board.
In machines I've built in the past (back in 1.0.0 days), config I'm
using and that card and 1.0.0 driver combo worked fine.
ztcfg reports no problems:
Yes, you could use MeetMe2 and MeetMe simultaneously.
~Vamsi
On Tue, 15 Mar 2005 08:01:28 +0900, Kuniyoshi Murata
[EMAIL PROTECTED] wrote:
Hi,
As I read http://www.areski.net/asterisk-meetme/about.php?s=0, meetme2
seems attractive to me. My question here is...
Can meetme2 and existing
Hello.
When the caller hangup the phone, asterisk kills my AGI python script without
notification.
I caught all signals, but none was trigered.
How can i trap this event to resume some operations.
Sorry for my poor english :)
Thanks.
___
On Wed, Mar 16, 2005 at 11:06:08AM +, Atif Rasheed wrote:
hello there,
I have searched lists about an application chan_spy, people talked about
it on lists that we can use it to monitor sip to sip calls. but I am
unable to find any clue of it.
can some one please tell me from where I
Hi Pepe,
You can't! As far as I can tell, once Asterisk eliminates an AGI upon
hangup, it doesn't send any signal information to the AGI script. If you
need to run some clean ups, the proper way to do so would be to execute
an AGI upon hangup, utilizing DeadAGI.
Nir S
-Original
On Tue, 2005-03-15 at 13:24 -0500, Erick Perez wrote:
im my case im looking into 100 seats initially and going up to 1000 at
the end (over a 18 months period).
Looks like we will have to develop *a lot* if we want to use * for it.
Maybe a commercial solution will be better at this time.
On
You can't! As far as I can tell, once Asterisk eliminates an AGI upon
hangup, it doesn't send any signal information to the AGI script. If you
need to run some clean ups, the proper way to do so would be to execute
an AGI upon hangup, utilizing DeadAGI.
You can also use FastAGI instead of
Hi Andrew,
thank you for the reply.
I have following your advice and I have put this into /etc/lilo.conf
append = pci=noacpi
Now proc/interrupts he returns me this:
[EMAIL PROTECTED]:~# cat /proc/interrupts
CPU0
0:1983298IO-APIC-edge timer
1:382IO-APIC-edge
Well,
It all depends what you want to do. We've already implemented a system
that can handle roughly 1000 channels of SIP using Asterisk. Of course we
used an Intel Cluster to reach that number, but the possibility exists.
It's all a question of design. I admit that using Asterisk would
You are correct, FastAGI is a valid option. However, if he's basing his
application on Asterisk Stable, FastAGI is not available in the stable
version.
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter
Sent: Wednesday, March 16, 2005
On Wed, 2005-03-16 at 11:38 +0100, Martijn van Oosterhout wrote:
Maybe it's been replaced by the Monitor app?
Or does it do something else?
The Monitor application records calls and writes wav files it does not
allow real time spying.
ChanSpy seems to have disappeared. The bug 2379 that
On Wed, 2005-03-16 at 13:08 +0200, Nir Simionovich wrote:
You are correct, FastAGI is a valid option. However, if he's basing his
application on Asterisk Stable, FastAGI is not available in the stable
version.
My version of Asterisk 1.0.6 includes FastAGI support and works pretty
well.
There was a thread some time back about making calls via * from a web
interface...ie user clicks number on web page and call is made...
There are basically two ways to implement this.
The first one assumes that your webserver is running on the same machine
as Asterisk. Then your web
Oops, you are correct, FastAgi is available in 1.0.6, my mistake
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter
Sent: Wednesday, March 16, 2005 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Those are the two valid methods. However, if you intend to generate many
calls, using the spool directory isn't a good method, as the spool is a very
slow means to do so. Using the manager proves more efficient for this task.
Nir S
-Original Message-
From: [EMAIL PROTECTED]
Thanks for that, I am however slightly concerned that due to the fast moving
asterisk project (with new versions coming out regularly) that digium may
start phasing out support for 2.4 kernel, I would like to settle on an OS
for my customers and don't want to have to readdress the situation in one
Hi all,
I have just published my last few weeks of hard work: IPSwitchBoard BETA.
Please let me know what you think and post comments on the Wiki.
http://www.voip-info.org/wiki-IPSwitchBoard+BETA
Thank you
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Hi!
I found some problems using the call forward.
I'm using this simple configuration, but something goes wrong, can someone
understand what is wrong and help me?
Thanks a lot
exten = _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2})
exten = _*5X.,2,Hangup
exten = *5,1,DBdel(CF/${CALLERIDNUM})
hello
i was searching for solution to problem (sip-h.323).
any one from this list asterisk mailing have any idea
how to fix it.
i am getting error when i try to call from sip to
h.323 user
i am successfully registering my asterisk box with
gnugk. but when i try to call to h.323 openphone on
On March 15, 2005 06:04 pm, Giudice, Salvatore wrote:
commercial licensing AND has a real enterprise class support structure
behind it, or are you going to run with PostgreSQL (bow wow) distributed
under a BSD license with some mom and pop support shops and some mailing
It's time to put up or
On March 16, 2005 05:57 am, pixer wrote:
I have following your advice and I have put this into /etc/lilo.conf
append = pci=noacpi
20: 0 IO-APIC-level t4xxp
modules (COM port, serial ports, etc), and shuffling the card around to a
different PCI slot, but unfortunately he does
Hi All,
I have been using kphone for quite some time and it
has been nice to me.I however wanted to know where (in
which files) and how is the STUN implemented in
kphone.
I am also trying to write my own software for a
softphone.Can anyone please giude me on how to
implement STUN in that taking
On Wed, 2005-03-16 at 11:20 +0100, Pepe Aracil wrote:
Hello.
When the caller hangup the phone, asterisk kills my AGI python script
without
notification.
I caught all signals, but none was trigered.
How can i trap this event to resume some operations.
Asterisk doesn't send any signal
On March 16, 2005 05:57 am, Andrew Kohlsmith wrote:
Can you put this card in a totally separate machine with your slackware
HDD
just to see if it comes up properly in another machine? This is very
unusual.
-A
Unfortunately I have already also tried this, without results.
I do not know
Hello!
I'm new to asterisk and because I try to configure the package for my
needs the last days without success, I'd like to ask a basical qestion.
I need asterisk to work together with the German VoIP provider sipgate
(http://www.sipgate.de). Asterisk should act as a softphone, I want to
im my case im looking into 100 seats initially and going up to 1000 at
the end (over a 18 months period).
Looks like we will have to develop *a lot* if we want to use * for it.
Maybe a commercial solution will be better at this time.
On Cebit SGI announced a server solution based on
Hi Ronald,
I have setup flash pannel, ... looks nice, but so far I could not
configure it to get more than 4x7 buttons.
I tried to make the buttons smaller, but than just the entire picture is
smaller.
What did you change in op_style.cfg? You can have literally hundred of
buttons per screen,
Hey there,
does anybody know a CLI SIP Client für Linux?
--
Mit freundlichen Grüßen
With kind regards
Klaus Peras
Support Networks/Networkmanagement
HOB GmbH Co KG
Schwadermühlstrasse 3
D-90556 Cadolzburg
Tel: 0 9103 - 715 -329
Fax: 0 9103 - 715 -299
Mobil: 0 175 63 78 911
URLs:
Hello!
Do somebody knows how to compile meetme2 with 1.0.6.
I readed wiki, applied patches, but no luck ;-(
Me be someone can give me working meetme2.c ?
:-)
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Hi Kamran,
Kamran Ahmad wrote:
hello
i was searching for solution to problem (sip-h.323).
any one from this list asterisk mailing have any idea
how to fix it.
i am getting error when i try to call from sip to
h.323 user
i am successfully registering my asterisk box with
gnugk. but when i try to
On Wed, 2005-03-16 at 12:15, Chris Blake wrote:
Greetings *`s,
There was a thread some time back about making calls via * from a web
interface...ie user clicks number on web page and call is made...
I`ve googled with a few words, checked the wiki, and tried to scan
through the archives,
Greetings *`s,
I have created a call file and am manually placing it in
/var/spool/asterisk/outgoing, but I get the following errors in the log
file :
===
Mar 16 15:26:57 DEBUG[2054]: Auto destroying call
'[EMAIL PROTECTED]'
Mar 16 15:27:43 WARNING[2054]: Unable to open
Hi,
I'd also like to see alternative op_style.cfg. Can we establish some place
to share them ? I've also one with smaller buttons (but will have to count
them :-) ...
Regards,
Rob.
- Original Message -
From: Nicolás Gudiño [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
I got the CVS head to compile finally, and yes I ditched odbc. noob or
not, it's a pain in the a$$ if you mess up the install. All in all,
mysql seems to work fine. Thanks.
Matt
-Original Message-
From: Joe Dennick [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 15, 2005 1:20 PM
Data validation should be done at all levels. Period.
Validating the SAME data at each level greatly decreases your speed.
True, but at the expense of data reliability and security. If one
validation layer is compromised (buffer overflow, packet injection, or
even a bad link between client
Hi,
Someone observed the problem with presence in
asterisk?
Please do reply.
With regards
Somesh S. Shanbhag
--- somesh s [EMAIL PROTECTED] wrote:
Hi,
I am again running with presence problem in
asterisk.
I have two windows messengers registered
successfully
with asterisk (Example msn1
On Tue, 2005-03-15 at 13:00 -0500, Giudice, Salvatore wrote:
MySQL: Speed, Power and Precision
_
Speed, yes. Anyone can write an SQL layer over a flat file and make it
fast. If you want real speed (faster than MySQL with the same level of
reliability choose
Hi everyone,
since I finished some hardware issues, now the real * configuration started.
It is my first attempt to get asterisk working and I am a bit confused.
The structure I am going to configure is quite easy:
The asterisk server is connected to a traditional PBX via S0.
When a user dial
What errors are you getting?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dmitry
Melekhov
Sent: Wednesday, March 16, 2005 4:36 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] meetme2 compilation
Hello!
Do somebody knows how to
An article posted on the The Register:
http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/
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Haha, yeah, plenty of hair left. I'm a youngin. Only 19. But that's
beside the point. If I had another box I could dedicate to asterisk, I
would do that without hesitation. Right now I just installed the new
Win32 version onto my dual booting XP/Debian laptop to play around and
get that set up as
On Wed, 2005-03-16 at 13:13 +0100, Christian Schoepplein wrote:
Hello!
I'm new to asterisk and because I try to configure the package for my
needs the last days without success, I'd like to ask a basical qestion.
I need asterisk to work together with the German VoIP provider sipgate
Chris Blake wrote :
-%-
If anyone can help I`ll send the call file to you, or is it ok to
clutter the list with it ?
-%-
'Clutter' the list I'd be interested and at least it is pertinent to *
;o)
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Hi everybody,
I'm receiving the message res_musiconhold.c:309 monmp3thread:
Request to schedule in the past?!?! in asterisk console when I try to
put a call on hold.
I don't the reason and I'm sure the relative module is loaded.
In musiconhold.conf I put these lines, trying something I
On Wed, 2005-03-16 at 14:20 +, Razza wrote:
Chris Blake wrote :
-%-
If anyone can help I`ll send the call file to you, or is it ok to
clutter the list with it ?
-%-
'Clutter' the list I'd be interested and at least it is pertinent to *
;o)
I am almost sure it has
Ronald Wiplinger wrote:
vpbx*CLI realtime update sippeers allow g729 name 621
Failed to update. Check the debug log for possible SQL related
That is the wrong format of the command. Notice the incorrect SQL that
was queried? Type realtime update by itself to see an example.
That is a
Hi,
I also wrote a PHP scripts that generate op_style.cfg. You specify how many
rows x cols and the icons/buttons/text alignment are properly scaled.
(i.e. you defined a 5 x 20 for 100 buttons, button height will be small so
Line, CallerID, Timer position will be adjusted)
Script not 100%
Nicolas,
I have setup flash pannel, ... looks nice, but so far I could not
configure it to get more than 4x7 buttons.
I tried to make the buttons smaller, but than just the entire picture is
smaller.
What did you change in op_style.cfg? You can have literally hundred of
buttons per screen,
Martijn van Oosterhout wrote:
On Wed, Mar 16, 2005 at 03:25:17PM +0800, Ronald Wiplinger wrote:
Mar 16 15:13:45 DEBUG[29502]: Raw Hangup 69.73.19.178:4569, src=14,
dst=1259 Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Update SQL:
UPDATE sip_buddies SET name = '621' WHERE allow = 'g729'
Mar
I have * running with sipgate.de so that works fine. However, if all
you want is to use * as a softphone, you'd be better off using an actual
softphone -- * would be overkill for that, and it still wouldn't be as
easy to use as a proper softphone.
-Original Message-
From: Christian
Thanks Steven, that was really a simple solution I overlooked. I added
appropriate context=siphones-superuser in the user settings in sip.conf,
commented out the includes under default and all inbound/outbound security
accounts are routed as I intended.
You were right, even unregistered SIP
Gianluca,
Did you install the .59r. Version of mpg123? The most common problem I
have seen for this is that people keep installing the 59q or 59g version
of mpg123. 59r is the way to go.
http://www.voip-info.org/wiki-mpg123
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
This Postgres vs. MySQL business is ultimately just a religious debate, like
PC vs. Mac, Ford vs. Chevy, or Kirk vs. Picard. They both work; they both
have their plusses and minuses; and debates about which are better never
convince anyone to change their preconceived ideas. It's also about as
Matthew Boehm wrote:
Ronald Wiplinger wrote:
[mysql1]
dsn = astconf
username = root
password = MyPassword
pre-connect = yes
You are not using the ODBC drivers. You can remove that [mysql1] stuff
from your res_mysql.conf
Removed, but still no codecs
br
*CLI Urgent handler
-- SIP Seeding peers from Astdb: '3044' at [EMAIL PROTECTED]:64718 for
120
Urgent handler
-- Registered SIP '3044' at 64.XX.XX.XX port 17524 expires 120
Codecs : 0x10c (ulaw|alaw|g729)
Codec Order : (g729|ulaw|alaw)
Using the following table:
CREATE TABLE
Dan, Thanks for the time helping me out. I figured everything out except
for the patch.
7. cd to asterisk/apps and run patch -p0
path-to/apps-meetme-cbmysql.txt
When I do this step it errors out and asks for the file to patch.. When I
look at the apps-meetme-cbmysql.txt It shows the file
I'm sure this is a pretty basic problem, unfortunately I am a telecomms rather
than a Linux person so any suggestions would be most appreciated. I have
successfully downloaded and installed the various Asterisk packages.
However, when I try to start Asterisk, I immediately get a message
Dear All,
I've setup got a Asterisk and pgSQL combi that works fine. I'm about to
perform the migration deployment when I noticed a issue which I need some
expert advise here.
When user connect to Voicemail, the CPU Load of the machine will shoot up
to around 50 - 60%, and its causing sound
On March 16, 2005 07:12 am, pixer wrote:
Unfortunately I have already also tried this, without results.
I do not know what to do any more..
Was it an entirely different motherboard (different manufacturer)? If so,
it's time to call Digium and open a ticket. It sounds like the card is DOA.
On Tue, 15 Mar 2005 14:13:40 -0500, Kanuri, Seshu (Company IT)
[EMAIL PROTECTED] wrote:
atxfer = *2 ; Attended transfer
Remove attended transfer capability and then you will be able o enter *2XXX
Jason
___
Asterisk-Users mailing
On Tue, Mar 15, 2005 at 08:38:04AM -0600, Matthew Boehm wrote:
I'm not a PRI expert and therefore don't know what this debug stuff means
for PRI, so if anyone can help me here...
I'm running the latest libpri and zaptel from CVS.
Keep in mind that everything works fine when using the STABLE
I installed this and it seems to be working great. Good job. Just one
question though, What is the shared extensions file?
- Original Message -
From: Thorben Jensen [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent:
Hi Ppl.
Once, couple weeks ago when I have updated * from CVS-HEAD something
happen and I could not send a fax anymore.
After that I have tried previous * CVS versions with different versions
of spandsp (0.0.1, 0.0.2pre4, 0.0.2pre10) but without any changes.
I have tried that on Fedora Core 2 with
Matthew Boehm wrote:
*CLI Urgent handler
-- SIP Seeding peers from Astdb: '3044' at [EMAIL PROTECTED]:64718 for
120
Urgent handler
-- Registered SIP '3044' at 64.XX.XX.XX port 17524 expires 120
Codecs : 0x10c (ulaw|alaw|g729)
Codec Order : (g729|ulaw|alaw)
Using the following table:
Hello:
Has anyone on this list had to configure hairpinning on a Cisco
gateway running IOS 12.2 or 12.3 and using a PRI for connectivity
to the PSTN? If so could you tell me how it is done? I'm told this
is the source of my call transfer problems and yet I cannot find
clear instructions for how
Hello:
Has anyone on this list had to configure hairpinning on a Cisco
gateway running IOS 12.2 or 12.3 and using a PRI for connectivity
to the PSTN? If so could you tell me how it is done? I'm told this
is the source of my call transfer problems and yet I cannot find
clear instructions for how
Hi Vladyslav,
Use 0.0.2pre1, but add the line
fax.verbose = TRUE;
just after
fax_init(fax, calling_party, NULL);
That will turn on the detailed logging.
Is the listing you posted the entire log? It looks like there should be
more.
One common mistake people make - Did you use the
Steve can you post your Cisco configs? Can you post the configs from your *
box that pertain to your issue?
- Original Message -
From: Steve Blair [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 16,
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af Henry Devito
Sendt: 16. marts 2005 16:17
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] IPSwitchBoard BETA
I installed this and it seems to be working great. Good job.
And what people are using to deploy super servers with astersik?
Itanium with linux? clusters of itanium with linux? or some RISC
processor with some *nix? cause it seems asterisk is only 100%
supported on Linux/Intel
or am i totally wrong?
On Wed, 16 Mar 2005 05:51:18 -0600, Rich Adamson
Erick Perez wrote:
And what people are using to deploy super servers with astersik?
Itanium with linux? clusters of itanium with linux? or some RISC
processor with some *nix? cause it seems asterisk is only 100%
supported on Linux/Intel
or am i totally wrong?
The highest-performing standard
Where did you get 1.05.23 from? The doc is available on the grandstream
site but not the actual firmware.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Tuesday, March 15, 2005 11:48 PM
To: asterisk-users@lists.digium.com
Subject: Re:
-- Goto (from-sip,901,7)
-- Executing SetVar(SIP/103-dfb6,
CALLFILENAME=20050316-181506-103-901) in new stack
-- Executing Monitor(SIP/103-dfb6, wav|20050316-181506-103-901)
in new stack
-- Executing Goto(SIP/103-dfb6, from-sip-post|901|1) in new
stack
-- Goto (from-sip-post,901,1
Thanks Kevin for this info,
If we want a box that can perform 60 calls. What would be apoproximate budget
for that using AMD x86-64 ?
µSelon Kevin P. Fleming [EMAIL PROTECTED]:
Erick Perez wrote:
And what people are using to deploy super servers with astersik?
Itanium with linux? clusters
(obviously if you do other magic in your dialplan this needs to be adjusted.
The important part is the 'g' flag to Dial (go on after hangup), and the NoOp
which echos the dialstatus and hangupcause variables to the console.
How would you do this in an AGI script? Basically what I have at
I changed the sequence first disallow and than allow. After
restarting * it is working now!
I am sure I copied the table and did not change it, ... somewhere it
must have the wrong order.
Thanks for your patient with me!
Glad we got it working.
-Matthew
Use whichever you want. Go get your own benchmarks. I'm sure you will
find benchmarks all over the web based on different conditions. The fact
remains that enterprises are deploying MySQL 4:1 over postergreSQL. I
believe the driving factors for this are the ability to commercially
license Mysql
to on
-- Executing GotoIf(SIP/103-dfb6, 1?7:9) in new stack
-- Goto (from-sip,901,7)
-- Executing SetVar(SIP/103-dfb6,
CALLFILENAME=20050316-181506-103-901) in new stack
-- Executing Monitor(SIP/103-dfb6, wav|20050316-181506-103-901)
in new stack
-- Executing Goto(SIP/103-dfb6, from-sip-post|901|1
[EMAIL PROTECTED] wrote:
If we want a box that can perform 60 calls. What would be apoproximate budget
for that using AMD x86-64 ?
60 calls can easily be done on a 3.4GHz Pentium 4 box, no special
hardware is required.
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How exactly does Asterisk provide E911 service??
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Once you run Dial from an AGI script, you lose control of the call via
the AGI script.
Jean-Michel Hiver wrote:
(obviously if you do other magic in your dialplan this needs to be
adjusted. The important part is the 'g' flag to Dial (go on after
hangup), and the NoOp which echos the
Klaus Peras wrote:
Hey there,
does anybody know a CLI SIP Client für Linux?
I think you may find one in Vovida.org
/O
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-Original Message-
From: Giudice, Salvatore [mailto:[EMAIL PROTECTED]
As for your 'artist license with your data' comment, put it into some
context. I would blame a programmer for trying to insert a
string of 255
characters into a field only 100 character wide. Maybe you could
On Wed, 2005-03-16 at 15:04 +, Andy and Jayne Slim wrote:
I'm sure this is a pretty basic problem, unfortunately I am a telecomms
rather
than a Linux person so any suggestions would be most appreciated. I have
successfully downloaded and installed the various Asterisk packages.
I believe the driving factors for this are the ability to commercially
license Mysql for product integration over PostgreSQL's BSD license,
This is a ridiculous FUD statement. Are you actually trying to suggest that
one cannot commercially license PostgreSQL?
That's simply FALSE.
The primary
Jason,
exten = s, 4, Dial(${VOICEPULSE}/011${ARG1}, ${LONGTIMEOUT}, Tt)
When I removed T and t options from dial command, the DTMF digit
recognition started working. Working line is below
exten = s, 2, Dial(${VOICEPULSE}/011${ARG1}, ${LONGTIMEOUT})
I will not change the features.conf, unless I
Hi,
I am playing with conferencing, but might have hit a bug... Any use who
wants to hang up or leave the conference should press the # key, after
which they get a goodbye message and the call gets disconnected.
However, this does not happen. whatever keys are pressed by whichever
party gets
-- DBget: set variable recv to on
-- Executing GotoIf(SIP/103-dfb6, 1?7:9) in new stack
-- Goto (from-sip,901,7)
-- Executing SetVar(SIP/103-dfb6,
CALLFILENAME=20050316-181506-103-901) in new stack
-- Executing Monitor(SIP/103-dfb6, wav|20050316-181506-103-901)
in new stack
How exactly does Asterisk provide E911 service??
It doesn't do anything with 911. You tell * what to do when someone
dials 911 via your dialplan.
To avoid legal issues down the road, I'd suggest handling it via a
local pstn line (one way or another), and install a Red Phone with
a normal pstn
On Wed, 2005-03-16 at 18:31, Kevin P. Fleming wrote:
[EMAIL PROTECTED] wrote:
If we want a box that can perform 60 calls. What would be apoproximate
budget
for that using AMD x86-64 ?
60 calls can easily be done on a 3.4GHz Pentium 4 box, no special
hardware is required.
Is that
Matt wrote:
How exactly does Asterisk provide E911 service??
Could you ask a slightly more open-ended and ambiguous question next
time? This one might actually have some real answers...
Asterisk does not provide _any_ service, the user configuring Asterisk
makes that happen. Asterisk can be
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