similar to: R: How to make * don't strip the leading 0

>> 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

Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following

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

Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called

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

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 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

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

This is super easy to do. All you need to do is to put that announcement in a MP3 and set a musiconhold class for that incoming Zap channel. Then basically when ever that PSTN number rings, Asterisk will play the MP3 stream "Your call may be monitored or recorded, please hangup if you do not agree...etc" in a loop until the line is answered. Caller doesn't pay a single dime to

> "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

>> 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 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

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

Hi folks! 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. So if I get a call from a mobile phone 0177-1234567 should be displayed, but 177-1234567 is displayed. I double checked if I've forgotten to remove an

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?

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 =>

-----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