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...eck sip messages. I have to sniff both asterisk and zoiper sip messages. I know that this can be tricky but this can help you to understand what is wrong in sip messages. Please let me know if you need more detail. Marino On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo < gincantalupo at fgasoftware.com> wrote: > Hi Marino, > > I tried to connect zoiper directly to the provider with the same account > parameters I'm using with Asterisk. Zoiper connects without problems. It > is true tnet.it is not resolvable but I can use the proxy URL > sip.tnet.it which seems to wo...

...box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at fgasoftware.com> wrote: > Hi Marino, > Asterisk gives a timeout on registration and a "no such host" because > cannot resolve tnet.it but that server address is not resolvable so I > think that is not a problem (my zoiper connects to the provider without > problems, so why shouldn...

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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

...les/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, does anybody know what *zttranscode *module* *is for*?* Thanks!! Giorgio -- _________________________________________________ Giorgio Incantalupo, mailto:gincantalupo@fgasoftware.com FG&A srl - http://www.fgasoftware.com - Voice@Work - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172

...is correct? Where can I find the drivers for this card? Should I re-compile linux kernel or there is another way to add drivers to linux? Thanks in advance. Giorgio. _________________________________________________ Giorgio Incantalupo, mailto:giorgio@fga-software.com FG&A srl (http://www.fgasoftware.com) - Formazione e Consulenza Tel: 02 9350-4780, 02 9350-0481, interno 104, Fax: 02 9350-0178 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041221/40971e48/attachment.htm

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 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

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

Hi, I'm making some tests with Tellfree brazilian provider. I'm using 2 users A and B, one for calling and the other to receive calls. When I make a call I can see (from the CLI console) user A is calling user B but user B does not answer (the phone continues to ring) even if the "sip show registry" command says user B is registered. In my sip.conf I have: register =>

...scussion > Subject: Re: [Asterisk-Users] how to show called name on > calling polycomdisplay > > I was looking for this exactly as well.... > > Any ideas? > > - Gabe > > > ----- Original Message ----- > From: "Giorgio Incantalupo" <gincantalupo@fgasoftware.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Wednesday, March 15, 2006 12:52 AM > Subject: [Asterisk-Users] how to show called name on calling > polycom display > > > > Hi, > &g...

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

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]:

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

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

...f [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'

...tuffed 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