similar to: Asterisk on Solaris 10

> 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

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

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

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?

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

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

>> 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

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

>> 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

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:

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

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

> 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

> Timeout, but no rule 't' in context 'home' > > about this line: > > exten => 2201,1,Dial(${PHONES1},20,Ttm) > > I know the problem is with the 't' but I don't know > what the parameters mean. I looking for a man page basically. The problem isn't related to the "t" in the Dial() command, which enables call transfer, but to a

Setup as follows: [private*] - Natting Router - [public*] [private*] cannot register via IAX2 correctly while [public*] is running. Status remains UNKNOWN even after minutes, calls from [public*] to [private*] are not possible. Console output of [public*]: | *CLI> iax2 show peers | Name/Username Host Mask Port Status | iaxtest/iaxtest (Unspecified) (D)

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 *

> 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

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

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

> "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