Displaying 20 results from an estimated 4000 matches similar to: "R: Asterisk Docs"
2004 Jun 24
6
R: How to force G729
>> allow=ulaw
>Why don't you remove this?
Because I need some other users to be able to call out using ULAW over the same PSTN gateway...
-Manuel
___________________________________________________
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com
2004 Jun 01
2
R: Hyperthreading?
That's the problem we had with Asterisk and HT on a 2.4 Kernel: whenever Asterisk was staying in the RTP stream, and HT was enabled (on a Dell Dual Xeon system), we had choppy audio. After disabling HT, everything was fine again. Nothing measurable, indeed, but you could definitely hear it. So there *must* be something.
-Manuel
-----Messaggio originale-----
Da: Peter Corlett
2004 May 18
1
G.729 on /dev/sda
I've just setup a new asterisk server, and I need to have G.729 working on this system. The problem is I don't have any IDE drives (and therefore no /dev/hda etc), but only /dev/sda.
Is there really *no* way to license G.729 on a SCSI-only system? IMHO it's really stupid to replace an entire server because of a licensing issue. There *must* be a solution.
Anyone, please? Or at
2004 Jul 12
1
R: How to make * don't strip the leading 0
> Is it possible to tell asterisk not to strip the leading 0
> of *incoming* MSNs? I use asterisk with i4l and whenever
> I get a call from an long-distance party, the leading 0, which
> should be there according the german numbering, is not.
Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped already by the phone company? I think the easiest
2004 Jun 18
2
cdr_addon_mysql compiling error
I'm trying to compile cdr_addon_mysql but keep getting this error. Again, searching the Wiki and the mailing list archive didn't bring up anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to switch back to 3.23?
# make
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c
cdr_addon_mysql.c:50: warning: parameter names
2004 Jul 07
1
res_odbc not working
I have been playing with res_odbc in these last days, but it doesn't want to work.
This is the output when starting Asterisk, so everything seems OK:
[res_odbc.so] => (ODBC Resource)
== Parsing '/etc/asterisk/res_odbc.conf': Found
Jul 7 20:11:32 NOTICE[-1084915040]: res_odbc.c:132 load_odbc_config: registered database handle 'mysql' dsn->[MySQL-asterisk]
Jul 7
2004 Jul 01
3
R: execute a context from cron
> I want to have call forwarding (from the POTS)
> turned on at the close of work and turned off
> automatically by *.
I would have a look at GotoIfTime:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
That should be much easier than a cron job
Regards
-Manuel
___________________________________________________
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax
2004 Jun 18
3
Thousands of contexts?
By reading the Wiki's I found out that an Asterisk server with many (>10000) extensions and/or SIP users can become slow when reloading. But what happens when you also have many contexts in extensions.conf? More precisely, one context for each SIP user?
I need this because I will have users with random usernames that they can choose, but I obviously cannot set that username as the outgoing
2004 Jun 22
2
Unable to find libiodbc.so.2
I was finally able to compile asterisk with cdr_odbc.so. But now for some reason I get that error:
*CLI> load cdr_odbc.so
Jun 22 16:38:53 WARNING[-1084309376]: loader.c:240 ast_load_resource: libiodbc.so.2: cannot open shared object file: No such file or directory
Unable to load module cdr_odbc.so
But the file is there...
# ls -lag /usr/local/lib/libiodbc.so*
lrwxrwxrwx 1 root
2004 Jul 07
1
Ringinbacktone even without 'r', and inexistant codec
I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk. It works ... Partially.
We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA.
First of all, when I make a SIP call to the unit with a simple Dial() command (no "r", so Asterisk shouldn't generate its ringback tone) I hear Asterisk's ringback tone anyway (I'm
2004 May 18
1
DateTime bug?
I've just checked out the latest CVS from the 1.0-stable branch, but DateTime() seems somewhat buggy. It says something like:
Tuesday May 18 11:46 AM 2004
instead of
Tuesday May 18th 2004 at 11:46 AM
(notice the wrong order of the words and the missing "th"/"at")
Did I miss something? Does DateTime() now take parameters that I wasn't aware of where you can tell *
2004 May 18
1
R: Configure asterisk for outgoing.. need authuser parameter?
Hi Tony,
Try adding "fromuser=xxxxx", maybe "username=xxxx" isn't enough... Just a guess, it already solved a few problems for me.
-Manuel
-----Messaggio originale-----
Da: Tony Hoyle [mailto:tmh@nodomain.org]
Inviato: martedì, 18. maggio 2004 13:03
A: asterisk-users@lists.digium.com
Oggetto: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?
2004 Jun 23
1
R: Which Linux ?
> Based on th wiki, avoid kernel 2.6 unless you know what you are doing.
> Likewise with fedora, which seems to work but needs kernel thread turned off.
Just my experience: I have installed Asterisk twice on Fedora Core 1 with kernel 2.4.22-1.2188.nptlsmp on Dual Xeon systems. It has worked perfectly both times, without needing any additional compiler flags, and no kernel panics.
What I
2004 Jun 21
0
R: Re: cdr_addon_mysql compiling error
Thanks for the tip, but adding the CFLAGS directive doesn't work either, same error message. I'll try to have a look in -dev, but if anyone comes up with a solution, a reply would be appreciated.
-Manuel
-----Messaggio originale-----
Da: Luckcuck Nick-LCKN001 [mailto:LCKN001@motorola.com]
Inviato: lunedì, 21. giugno 2004 13:52
A: asterisk-users@lists.digium.com
Oggetto: RE:
2004 Jun 21
1
R: Re: cdr_addon_mysql compiling error
>> I'm trying to compile cdr_addon_mysql but keep getting this error.
>> Again, searching the Wiki and the mailing list archive didn't bring up
>> anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to
>> switch back to 3.23?
>>
>>
>> # make
>> cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o
2004 May 24
1
extensions/sip from database?
We are planning to deploy a pretty large asterisk server with many SIP extensions (might be up to 10000 in the future), and I have a few questions:
1) is this possible, or are we running into some kind of limitation in the software that I wasn't aware of and that I didn't find by browsing through the archives and through Wiki? No, we don't need any G729-G711 transformations, it would
2004 Jun 24
2
How to force G729
We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729.
The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that:
[mypstngate]
type=friend
2004 Jun 24
2
R: R: R: How to force G729
> "If" I understood your initial objective correctly (and I may not have),
> the user's phones are negotiating the codec to be used for each rtp session.
>
> Asterisk parameters can be used to dictate rtp sessions between the sip
> phone and asterisk, but that won't influence the next step in which the sip
> phone negotiates a new rtp session directly with the
2004 Jun 16
0
Disable authentication on outgoing SIP calls
I am trying to make Asterisk communicate with a voice switch which doesn't need (and like) authentication on outgoing SIP calls. I have configured it as follows in my sip.conf:
[myswitch]
type=friend
host=192.168.1.100
port=5060
context=default
canreinvite=no
To dial out using this switch (it acts as a PSTN gateway) I use this in extensions.conf:
exten =>
2004 Jun 18
0
R: Thousands of contexts?
-----Messaggio originale-----
Da: Kevin Walsh [mailto:kevin@cursor.biz]
> I don't quite understand your Caller*ID dilemma.
> In your sip.conf, you'd have a block for each user, say [abc123].
> That's your random username, yes? The same block would also
> define the password and other directives. Why can't you simply
> include the "callerid" directive