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...and dial multiple phones/lines simultaneously. If I use this Dial command: exten => 222,2,Dial(${TRUNKBP}/246&SIP/258&${TRUNKBP}/243,20,tTr) ... all phones ring just once, after that only the first one continues ringing and only that one can answer. Can anyone tell me why? thanks! Tomica -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040901/5a60afc9/attachment.htm

...my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. The database is created, cdr table also, the username and password is right. I have tried configuring cdr_mysql.conf to connect via localhost mysql.sock or via tcp port, but in both cases I got this error. Thanks! Tomica Crnek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040202/49c83d04/attachment.htm

Hi everyone, Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. Tomica ---- This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr

...--------------------------------- -- Executing Dial("SIP/7001-da6d", "Zap/g1/098227655") in new stack Jan 31 17:36:20 NOTICE[1256866752]: app_dial.c:527 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time an anyone help, please! Tomica Crnek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040131/95a511d2/attachment.htm

has anyone seen this? -- Starting Skinny session from 192.168.13.102 -- Starting Skinny session from 192.168.13.102 triton*CLI> Oct 15 13:44:05 WARNING[213019]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected. Oct 15 13:44:05 NOTICE[213019]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Success Oct 15 13:44:05

...nk, to PSTN and if the called phone is busy I don't hear anything!?! I should hear tone indicating that called number is busy. I have played with busydetect and callprogress in zapata.conf, but I didn't find what is wrong. I would appreciate if someone can tell me what to do. Thanks! Tomica Crnek --------------------------------------- tel: +385 1 6690200 dir: +385 1 6690250 gsm: +385 98 227655 SPAN d.o.o., Turinina 3, 10000 Zagreb, Croatia -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/...

...gt; --__--__-- > > Message: 1 > From: "Kevin Walsh" <kevin@cursor.biz> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] Caller ID via IAX > Date: Mon, 12 Apr 2004 01:30:40 +0100 > Reply-To: asterisk-users@lists.digium.com > > Tomica Crnek [Tomica.Crnek@span.hr] wrote: > > (Article auto-converted from unnecessary HTML to > nice plain text.) > > > > I have two Asterisk boxes connected via IAX. I > would like to transfer the > > original caller ID from one to another but I > always get "Guest...

Hi All Has anything been done to fix the issue where the * box is sat behind a nat firewall? Regards Mark

Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike

Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone

Hi, I am having some problems mentioned below, the box is in production live environment with traffic around 30 - 100 calls. I am running T/E410P in a Dual P4 xeon with HT disabled. I am using zaptel 0.9.0 and asterisk stable 1 release. There is no gui, just mysql, perl (small script) and asterisk. System runs very smoothly if the calls are around 40-50 and comes one by one , however sometimes