search for: _fangi_

...pe switchtype=euroisdn overlapdial=yes pridialplan = unknown prilocaldialplan = unknown nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=no group=1 language=it musiconhold=default channel => 1-15,17-31 Thanks for help, _fangi_

...g till it receives a hangup signal. I think this is due to the fact that the message is played without the call has been answered, but I'm wondering if there's some way to let Asterisk realize it. All I see in the CLI is the line "PROGRESS with cause code 0 received". Thank you, _fangi_ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060227/049e5433/attachment.htm

...right and bristuffed zaptel module for TDM40B works right. The card does not share IRQ with other devices, anyway I tried to have only TE110P mounted on PCI slot and to change PCI slot where card is mounted. Nothing to do. I really don't know what else I can try. Thanks for help, _fangi_ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051012/11cc8cb6/attachment.htm

...actually applies cleanly to 1.2 . >Contact me by email for packages. Though I suspect that it is not going >to solve the problem. Patched zaptel 1.2.0beta1 with bristuff 0.2.0-RC8o and now modules load and unload well. Now have to try Asterisk, but this is another story... Thanks for help, _fangi_

...e responds, then bridge the two calls. What I can't manage is jump to next dialplan command soon after callee enters the queue in order to call other numbers. I also tried AGI and Asterisk Manager, with the same result. I think I'd need some kind of multi-threading. Any ideas? Thanks, _fangi_ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060220/3109ec6e/attachment.htm

That's exactly what I was looking for. By the way, I discovered Local channels to fork into dialplan. I also discovered that roundrobin policy does not work as I expected, but that's another story. Thanks for help, _fangi_ -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Chris Bagnall Inviato: luned? 20 febbraio 2006 20.21 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users] queue beh...

...any unreachable mobile from Fastweb or Telecom PSTN I don't get any courtesy message. The same happens when calling an inexistent number. I'm configuring two PBX's, connected to two different phone lines, both behave this way. Perhaps there's some missing zapata parameter? Regards, _fangi_ > Well, > > it's funny because here, now (Italy; Telecom Italia PSTN calling Wind > mobile), I do get the courtesy message saying that they're moving me to > voicemail, if I call myself from the office PBX to my mobile Wind > number, and the cellphone is switched off.

...only their cards can work with their drivers. In the same document and googling around I found that bristuff source files must be patched to recognize other cards. Anyone has experienced with this? The alternative would be using mISDN, but I'm not sure it's as fine as bristuff. Thanks, _fangi_ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060322/bbacb966/attachment.htm