similar to: cannot get calls with Tellfree brazilian provider

Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not

Hi, Yesterday I upgraded one of our system from 2.0.7 to 2.0.9, I've got about 5 panic since then. It is using the exact same config (see attached) and was compiled the same way. Any clue on the reason? And how can I fix that? Jan 17 12:06:20 server dovecot: imap(XXXX at YYYYYYY): Panic: file imap-client.c: line 570 (client_continue_pending_input): assertion failed:

Hello I was trying to figure out how to add printer drivers manually for downloading to Windows 2000 clients and I came accross this problem. (The printer drivers were installed some time ago using an earlier version of samba 2.2.x). $ dbx /usr/local/samba/bin/rpcclient (dbx) run xxxxxxx -U yyyyyyy Running: rpcclient xxxxxxx -U yyyyyyy (process id 4735) Password: rpcclient $> enumprinters

Hi All, I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX 2.7.0.10. I have configured x100p fxo card in my asterisk box. But in my cli mode i am not getting the command *"dahdi show status"* Output of CLI : astrisks*CLI> *dahdi show status* No such command 'dahdi show status' (type 'core show help dahdi show' for other possible commands) I

[Since samba-bugs has been shutdown, I'm posting this here...] I'm running RH 7.0 with both 2.2.16 and 2.4.2 kernels I've got the following samba packages installed: samba-client-2.0.7-21ssl samba-common-2.0.7-21ssl I have a Windows NT 4 server (SP6a) which stores many of my files since I use both Linux and Windows to access them (Linux on my desktop, Windows on my laptop) I

Hi, Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help

I have been successfully connected (incoming and outgoing) to FWD for a very long time. A few months ago, I changed from SIP-based FWD service to IAX2-based, and that went fine as well, both incoming and outgoing. At the time, I was running Asterisk 1.0.3 Stable. I rarely use the service, so other than noticing that I was always successfully registered to FWD, I didn't make or receive calls

Dear UseRs, I'm having trouble drawing a barchart with 2 quantities in different units as follows with one unit on the top axis and the other unit on the bottom. E.g., x Tonnes | . . . Row 1 Tonnes1 | xxxxxxxxxx Row 1 Litres1 | yyyyyyy Row 2 Tonnes1 | xxxxxxxxxxxxxxxxxxxxx Row 2 Litres1 | yyyyyyyyyyyyyyyy

Hello. I've setup a FreeBSD 4.x machine with Samba 2.x and and OpenLDAP backend. I recently upgraded to Samba 3.0.10 and put "passdb backend=ldapsam_compat" in smb.conf. Now, everythings seems to work all right, but I see a lot of this entries in the logs: Feb 17 19:42:27 xxxxx smbd[44170]: get_alias_user_groups: gid of user yyyyyyy doesn't exist. Check your /etc/passwd

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

Hi to all, I and using asterisk with following 1. TDM400p card with four FXS modules, So there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 But when some dials form his number (suppose abc) to my number (suppose xxxx) I get abc number on my analog phone, but now I have purchased more than one numbers

Package: xen-api Tags: l10n patch Severity: wishlist Hello, Please, Could you update the Brazilian Portuguese Translation? Attached you will find the file pt_BR.po. It is UTF-8 encoded and it is tested with msgfmt and podebconf-display-po. Kind regards. -------------- next part -------------- A non-text attachment was scrubbed... Name: pt_BR.po.gz Type: application/x-gzip Size: 958 bytes Desc:

Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

Dear all I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on is asterisk and it is working fine but i got this DTMF error on asterisk CLI what is it ?? -- Zap/36-1 is ringing -- Zap/36-1 answered SIP/5406-9fa59770 -- Channel 0/1, span 2 got hangup request, cause 31 [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to forward voice or

Help ! I am very happy with my 20001026 wine, running my favourites windows applications. Unfortunately, I still have a minor problem: My keyboard in perfectly recognized as the Brazilian ABNT layout, I had to change something in the keyboard.c file, but now is ok. So when I start my application I can get every character I need, but I can tell windows that a '~' followed by a