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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...

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

...? I think 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" <aste...

...? I think 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" <aste...

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

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

...s. Yuck. By 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...

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

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

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

Hello, I just did a fresh svn install of 1.4 trunk everything. Everything compiles and installs just fine. When I get to asterisk-addons, I cannot select chan_mobile in menuselect. Chan_mobile is not even an option in menuselect for asterisk trunk. I tried the latest patch which failed in many places but did add an option for chan_mobile in menuselect for asterisk but it still cannot be

I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with

...e programmable 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 t...

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

has anyone experienced a problem where RTP audio cuts out when doing 30-40 concurrent channels via sip? The box is freebsd 6, asterisk 1.2.4, voip only (no libpri, no zaptel - not even a timing source) The box has plenty of bandwidth, when a call to the same box is iax2 it works, but when its sip a call gets connected a few frames of audio are passed and then silence. When the box is completly

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Any ideas? Sangoma support is closed for the evening. I have the latest Sangoma drivers and Asterisk 1.4 everything installed. When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up" repeated over and over. The B channels never come up. There are no errors in any of the logs, zttool, or the wanpipe tools. Intense pri debug output: < Unnumbered frame: < SAPI: