similar to: Disable authentication on outgoing SIP calls

>> 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

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

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

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 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

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 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

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

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'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 *

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?

-----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

If I have ODBC logging enabled (with cdr_odbc), Asterisk logs everything to ODBC *and* to the CSV file (Master.csv). If I issue a "reload", it stops logging to the CSV file, but continues logging to ODBC.   To have it log to the CSV file again, I have to issue "unload cdr_csv.so" then "load cdr_csv.so".   Is that normal behaviour? Is it supposed to log to the CSV file

I have defined a SIP friend without username and secret, only IP-based. I have also defined an accountcode for that "friend", as follows:   [mypeer] type=friend host=192.168.0.100 port=5060 context=mycontext canreinvite=no accountcode=mypeer   Unfortunately the accountcode for the calls originating from "mypeer" doesn't show up in the log (either CSV or ODBC). All the other

> 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

When a user calling over the PSTN network calls one of our SIP users with a restricted number (CLIR), our PSTN gateway is sending us incoming calls with the following additional headers: Proxy-Require: privacy Anonymity: uri Remote-Party-ID: <sip:41781234567@192.168.0.100:4000>;privacy=uri as opposed to Remote-Party-ID: <sip:41781234567@192.168.0.100:4000>;privacy=off when CLIP

>Define that per user. Of course... The user part is not the problem. If I force a user in its extensions to use G729 only, he actually talks G729 to Asterisk, but asterisk still talks ULAW to the PSTN gateway, doing the transcoding. This is driving me crazy... -Manuel ___________________________________________________ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax

> 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

Does anyone know of a software SIP fax client? Something I can install on a PC which connects to the asterisk server and sends/receives faxes? Something like XLite - but to fax instead of to phone.   I know of the "fax machine connected to an ATA" solution, but that's not really what I'm looking for :-)   Thanks -Manuel   ___________________________________________________