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hot to asterisk multiport...??? example 5060, 5061, 5080 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070815/8a515309/attachment.htm

Hello: Have a TE110P laying around and decided to see if I could build a router around it. I've tried compiling several versions of zaptel .1.4.x with the same results. I checked the zaptel changelog and can't find anything about it no longer being supported (or that it ever was for that matter). ztcfg: Zaptel Configuration SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > Are you using the wifi on on the cellphone? The peer IP is showing as > 192.168.200.3 which is not a routable address. Unless things have changed, > double NAT configurations do not work. Hi Steve, My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direc...

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

...there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08, Steve Totaro <stotaro at totarotechnologies.com> wrote: > From: Steve Totaro <stotaro at totarotechnologies.com> > Subject: Re: zap not coming online on fedora 8 > To: "bilal ghayyad" <bilmar_gh at yahoo.com>, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists...

...there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08, Steve Totaro <stotaro at totarotechnologies.com> wrote: > From: Steve Totaro <stotaro at totarotechnologies.com> > Subject: Re: zap not coming online on fedora 8 > To: "bilal ghayyad" <bilmar_gh at yahoo.com>, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists...

I need your help: please help test the gender detection module at 575-613-4392. I wrote a gender detection module and thought I'd try it out. It only takes a second. I've been showing 90%+ accuracy and I want to make sure it's working correctly. Rain and significant background noise seems to throw it off, so I still have a bit of work to do. Have your friends and significant others

Hi all, Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

...the time you code > in logic for handling timeouts and incorrect responses to menu's with > all the gotos and what-not, it turns into a god aweful mess. > > Sure, you can do it. > > Doug. > > > > ----- Original Message ---- > From: Steve Totaro <stotaro at totarotechnologies.com> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Sent: Thursday, July 10, 2008 10:37:55 AM > Subject: Re: [asterisk-users] Asterisk as an IVR solution > > > > On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpen...

I can ping it just fine. I am on gw5.voicepulse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040615/054c83f4/attachment.htm

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra

...stall in the correct order. Try the awesome asterisk_update.sh shell script. Are you trying to emulate CPE or NET? Try signalling=pri_cpe Check for whitespace behind the statement, zapata.conf seems bitchy about whitespace. hth -----Original Message----- From: Steve Totaro [mailto:asterisk@totarotechnologies.com] Sent: Wednesday, September 28, 2005 12:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net' Any ideas? 51] logger.c: [chan_zap.so] => (Zapata Telephony) Sep 28...

Since it is Saturday evening (7PM EST) I am asking this on the list in case someone who knows sees it and can answer. Astricon is in my back yard for the first time, and I could hit you with a rock. I would always like to attend, and spoke at the 2007 Astricon in Phoenix but don't have the idle cycles. Question: Can I just go to Astricon and take the dCAP exam only? In and out? Cost? I

...ble by user. Questions: 1. Need to know the cpu that can support asterisk for this type of application 2 What else do we need on the box to support the application 3 Any pointers related to this would be really appreciated. Thanks Ram On Mon, Oct 13, 2008 at 4:55 PM, Steve Totaro < stotaro at totarotechnologies.com> wrote: > > > On Mon, Oct 13, 2008 at 5:53 PM, Steve Totaro < > stotaro at totarotechnologies.com> wrote: > >> >> >> On Mon, Oct 13, 2008 at 5:22 PM, Kevin DeGraaf <kevin at kdegraaf.net>wrote: >> >>> I need to monitor the state...

Unbeatable price for a low end Asterisk server (or any server for that matter) http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l=en&oc=bednv4k&s=bsd I wonder if anyone has any experience with this box and Digium or Sangoma hardware? Any compatibility issues? If not, I might stock up on them. Thanks, Steve Totaro -------------- next part

So we've got a TE410P configured as E-1. The PRI is showing up as normal, I have green lights, but d channel doesnt seem to come up and i keep getting this error if i do a "pri intense debug" The carrier swears up and down that there are no issues on their end. Any thoughts? localhost*CLI> > Unnumbered frame: > SAPI: 00 C/R: 0 EA: 0 > TEI: 000 EA: 1 > M3:

Hi all: Does somebody know how many meters of cable is supported between Asterisk Digium-FXO cards and the analog telephone ? Thanks, Angel

...n infinite number of calls (meaning: maxlen=0/no limit), and the agents are already answering the calls immediately/after one ring, but the problem still shows up. Is forwarding to a meetme conference room faster than through a queue? On Thu, Oct 16, 2008 at 11:25 PM, Steve Totaro < stotaro at totarotechnologies.com> wrote: > If you can figure out how to generate .call files from your DB > entries, you have it made. > > Vicidial needs alot of work as far as I am concerned, for free it is > OK I guess. I think using meetme conference rooms for everything is a > kludgy hack, and the...

hello to all, I advice you to not use CDRtool from ag-projects : Fisrt ag-projects talk about is product like a gpl software however they don't provide at least some documentation for non commercial users . try to call them !! i'll offer you some money . You can not Call them for some advices ... It's really a bad product don't waste your time to setup it. this enterprise must