Displaying 20 results from an estimated 10000 matches similar to: "Realtime: Should I say or should I go (now) ?"
2007 Oct 25
3
Realtime on Asterisk 1.2.24
Does realtime work reliably on Asterisk 1.2.24?
Are there any definitive guides, I can only find bits and pieces here
and there. Any accurate howtos would be of great help.
I am missing func_realtime.so. Where does this file come from?
Asterisk or asterisk-addons? I saw in one of the howtos that it is
needed. Is it needed for 1.2.X or 1.4.X. Also, what about the switch
lines in the
2007 Sep 04
11
stop log/debug messages into /var/log/messages
Hello good ppl,
Any way of stopping asterisk writing into syslogs or any other file, if
I set verbose 6 on the console?
All I want is the verbose output only on the console, nowhere else.
My logger.conf says :
console=> notice,error
;messages => notice,warning,error
Thanks in advance.
- Benjamin Jacob.
EMAIL DISCLAIMER : This email and any files transmitted with it are confidential
2007 Dec 17
3
VoIP service providers/PSTN termination points
Hello ppl,
Am looking at some PSTN termination providers in US. If this question
has been repeated, please point me to the correct link, as I've tried
searching the archives but have been unsuccesful so far.
I have come across quite a few companies which provide the same, such as :
Iconnecthere <http://www.iconnecthere.com>
Vonage <http://www.vonage.com>
Teliax
2007 Aug 20
1
1.4.4. caller ID not working ?
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten => _0.,1,Set(CALLERID(all)=Ben Jacob <988077>)
exten => _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set(CALLERID(all) above supposed to change the
display name as above(Ben Jacob) and change the From URI to 988077 at myip??
As of now, only the _display name_ is being replaced, but not the
2008 Jul 01
3
music on hold realtime
Hi,
Is it possible to use realtime for Music On Hold?
Is it also possible to store the music/audio files on the database, same
way a voicemail can be stored on the database?
Thank You
Regards,
Nhadie
2007 Nov 22
1
common/shared voicemail box
Hello All,
I am using ODBC storage for voicemail on my asterisk box. I want to have
a common voicemail box for different extensions.
I know how to do that, but the question troubling me is how and where do
I store the the extension name for which a particular voicemail was left.
e.g. extensions 1000, 1001, 1002 all using the same voicemailbox 55555.
Now, when someone calls 1000, and leaves a
2009 Feb 25
1
Realtime database function help
Hello Everyone!
According to voip-info.org the correcy syntax for the realtime function is:
REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read
REALTIME(family|fieldmatch|value|field) on write
It seems from the syntax that it is only possible to retrieve a full
row according to the value of only of column. This translates in SQL
language as Select * from family where fieldmath =
2008 Jan 10
1
Asterisk Realtime unixODBC timeout?
How does one get asterisk to timeout realtime request via res_odbc to
unixODBC? I've set timeouts as appropriate for freetds (which
unixODBC is using.) However, it doesn't seem to work. It takes over 3
minutes to timeout a connection and queries never seem to timeout, so
a channel waiting on a query never terminates.
I did notice that res_odbc.c never sets a timeout on the query
2008 Mar 08
3
replace astdb with a cluster-capable sql database engine
I've been searching the Internet for information
regarding the replacement of astdb with a modern sql
engine.
There are several reasons one would like to do this.
First of all, external applications have a hard time
reading/writing to the now-old astdb format.
Also (and this is what interests me most), the sql
astdb could easily be clustered throughout several
servers (I'm looking for a
2007 Aug 01
2
multiple pbxes, multiple domains, same user ids?
Hello good ppl,
A couple of questions for multiple pbxes
1. Is it possible to support multiple pbxes in one Asterisk box(using
contexts, etc.)?
2. Can we use the "domain" field in sip.conf to specify the different
domains for sip users, having one domain for each pbx?
I just tried registering two xlites, with different domain names (with
the same specified in sip.conf). But, Asterisk
2007 Nov 28
2
Billing/Call Control engine : AGI scripts/ AstMan API
Hello ppl,
Have implemented a really nice Billing engine using AGI scripts. So far
it works fine, tho haven't yet put it in the torture cell.
The AGI scripts have been written in PHP, using MySQL for the billing
and profile information.
The major disadvantages I see using AGI scripts :
1. A new process(invocation of PHP scripts) on every new call.
2. MySQL connections on every instance of
2008 Oct 08
1
make func_realtime work like app_realtime (1.6)
Yell at me if you will, but I hate func_realtime - it's not very usable nor
is it change-friendly (update your database and your dialplan completely
breaks).
I'm getting a new 1.6 box built out and working, and wanted to emulate the
functionality of APP_realtime somehow, so I started digging around in the
func_realtime source - here's what I came up with:
For 1.6.0, look at line 86
2007 Sep 18
4
Linux limits
Hi all,
Any one know how to increase the Linux limit? I am hiting a wall on 200
calls playing files at the same time. From Asterisk console, I am
getting messages like
Sip_request_call: Unable to build sip pvt data for "asterisk1/700"
Too many open files
Is this a limit of my Linux box? I only have 512MB of ram. Will increase
it to 2G help or I have to change some configuration in
2008 Sep 08
2
Pointers to replace astdb
Hi listers,
We want to implement one call center with asterisk. The idea is it should be
scalable, with openser as an dispatcher and bunch of asterisk servers to do
ACD, Queues, Agents things... Easy to say :(
Look closely to the current asterisk, we do see some problem:
- SIP registrations was stored in astdb.
- And queue members also was stored in astdb.
- ...
asterisk was built as
2009 Mar 21
2
1.6.2 beta 1 crash
Hi,
I'm starting testing 1.6.2 beta. CentOs 5.2
I found my first crash, first I have
[Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql:
Attempted to update column 'useragent' in table 'sip', but column does not
exist!
[Mar 20 20:30:41] ERROR[11201]: res_config_mysql.c:581 update_mysql: MySQL
RealTime: Updating on column 'lastms', but
2018 Dec 06
3
how to use a database
On 12/05/2018 05:00 PM, Antony Stone wrote:
> On Wednesday 05 December 2018 at 15:31:38, hw wrote:
>> I don't see a table for that.
>
> You need to create that for yourself.
>
> Asterisk can write to database tables, but doesn't help you set them up, for
> reasons I can't comment on.
How do I know what the schema needs to be? Does anybody have a scheme
for
2011 May 11
4
concurrent call tracking
Hi all,
I would like to track/store concurrent call usage per user by
day/week/month and get server totals by day/week/month. Google comes up with
mostly info regarding concurrent call limits, though my goal is to calculate
actual concurrent channel usage and add it into reporting. I'm using * 1.6.2
+ mysql - realtime (no gui). Any suggestions / open-source / AGI on where to
start looking
2015 Jan 25
2
Wiki (pjsip+realtime) says don't put the transports into realtime. Still true?
Hi,
The asterisk wiki page says:
"Sorcery.conf allows you to try to configure other PJSIP objects such as
transport using realtime and it currently won't stop you from doing so.
However, some of these object types should not be used with realtime and
this can lead to errant behavior."
Which objects and is this still true in 1.13.1 ?
Thanks,
Antonio.
PS:
2011 May 23
1
[Fwd: FW: extconfig.conf]
Hi Andrew,
OK, (the simple fact that those machines are not connected to internet
makes that i have to go to those machines and copy them on a usb-stick,
so it causes some delay each time...)
-------- Forwarded Message --------
Sorry - I meant extconfig.conf - not cdr_mysql.conf (my mistake).
I use (and done for a long time) mySQL for realtime storage - and it's
never let me down (touch
2015 Jan 29
2
any valid up-to-date info about Kamailio-Asterisk integration ?
Hi all
Have recently watched Matt Jordan's session on Kamailio World 2014
On slides 26-29 of his presentation
(http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf)
he speaks about a (completely new, for me at least) approach to build
scalable telephony systems, using N instances of Kamailio and N
instances of Asterisk
Are there any