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...od: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format* Is there anybody who can help me?? TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

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

...Speex-dev mailing list > > Speex-dev at xiph.org > > http://lists.xiph.org/mailman/listinfo/speex-dev > > -- > Randy Yates % "She's sweet on Wagner-I think she'd die > for Beethoven. > Digital Signal Labs % She love the way Puccini lays down a > tune, and > mailto://yates at ieee.org % Verdi's always creepin' from her > room." > http://www.digitalsignallabs.com % "Rockaria", *A New World Record*, ELO > -------------- next part -------------- An HTML attachment was scrubbed... UR...

I have a need to have the two sip phones register with the same extension (at least I think I have the need :) A client wants an incoming call to ring at the receptionists desk and also at their desk. If the receptionist is in it will be answered there and put on hold followed by a "Joe, you have a call on line 1". Is there a way to do this w/ asterisk? I've played with two

Hi, I am currently working on C5505 USB stick http://focus.ti.com/docs/toolsw/folders/print/tmdx5505ezdsp.html to sample input voice and encode it. For compiling and buring usual programs, I am using CCStudio 4.0. For encoding voice samples, I am using Speex codec binary http://downloads.xiph.org/releases/speex/speex-1.2beta3-win32.zip or

Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models?

...____________________________ > Speex-dev mailing list > Speex-dev at xiph.org > http://lists.xiph.org/mailman/listinfo/speex-dev -- Randy Yates % "She's sweet on Wagner-I think she'd die for Beethoven. Digital Signal Labs % She love the way Puccini lays down a tune, and mailto://yates at ieee.org % Verdi's always creepin' from her room." http://www.digitalsignallabs.com % "Rockaria", *A New World Record*, ELO

Hi, in a VoIP application, the endpoint A send speex payload to B. B doesn't know how A acquire audio, it only know that the channel is narrowband so, how can B know the size of the output buffer to pass to the speex_decode()? Thanks, Daniele.

hi, i am trying to do autoattendant but failing. as in the manual i inserted the background(welcome-mainmenu) file so that after the sound the caller can dial the extension he wants to call. i figured that the background sound wasn't coming in the asterisk. how do we do this without first loading the welcome message? for example after certain rings the caller can dial the extension no to

...ote that if I put the two lines of [ins_ext] context after "exten => 103,2,Hangup" extensions 104 can be called) TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid

While on conversation with another party, I dial the atxfer key sequence. Asterisk says "Transfer" then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep'

...I'm using Asterisk 1.0.7 (bristuffed but I don't think this is the problem). Can you (or somebody else) help me please? TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

I often get a Maximum retries error while making outgoing calls. Why does this happend? Most of the time a reload solves the problem, but not all the time? What to do? Aug 26 09:52:46 WARNING[6613]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 32166-1B75-151E-E6DA-E37AEEAA2882@10.100.4.252 for seqno 1 (Non-critical Response) Regards, Arne Morten.

...o/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-user...

...ere anybody who knows what this warning means?? WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

...it appears when typing lsmod) nothing has changed. Should I buy a x100p to get the right timing? Or there is another solution? TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

...r turns off, my telephone connection to Fastweb drops (while internet connectior is ok)? Restarting Asterisk is worth nothing. TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

Hello, I am trying to setup my Asterisk box to accept calls from the FWD network. I've followed all the config advice / samples I've found on the web. Making calls to devices on the FWD network from my Asterisk box works flawlessly, but whenever I try to call my Asterisk box from a FWD client I get a busy signal, and a "Call Disconnected" 486 error. What's odd is that I

I cannot reach voip-info - is it just me or is the site not available ? Julian