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Hi, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, is there anything going with VoiceXML in asterisk??? is this the list to query regarding this or should I put this on the dev list? thanks, dave cantera

I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the

I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated.

Dave, Docos for Comedian Mail? Justin >From: dave cantera <david.cantera at iacnet.net> >Subject: Re: [asterisk-users] Need good voicemail documentation > >An HTML attachment was scrubbed... >URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080208/501668f8/attachment.htm _______________________________________________...

Dear all, I have two system Sys A and Sys X. Sys A is normal PC. Sys X have installed asterisk 1.6 and i want register(or reserved) sip extension(like 4001,4002,4003..) through Sys A(Sys A have some ip address) but i don't use any soft-phone means i want to write Perl or php(any language) script to register sip extension. Suppose to 4001 is reserved with Sys A. 4002 is reserved with

hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070320/227a3b32/attachment.htm

Hi Greeting to all you out there. I am new at asterisk, I have been working with PLMN platforms telecommunication for 5 years with NSN and Huawei. We have recently built an asterisk PBX with Trixbox and connected it to our MSS using Digium E1 cards(ISDN). Everything went smoothly as there are tons of information out there, except for the TON number. If you have worked in Telecommunication you

...tion and answer machine detection get detected as the same. If i need to seperate the two how do i do that? For example, if i use AMD() to detect an answer machine by saying any greeting exceeding 2.5 seconds is a machine, how do i distinguish between a fax/modem and a long greeting? ---- dave cantera <david.cantera at iacnet.net> wrote: > carlos, > you got further than I did... AMD didn't work at all on my release.. I > think I was using 1.4.11 at the time... > I ended up using the below > daveC > > ;-----------< amdtest (ext 13) starts here >------------...

This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many

...needs, send screen pop to a remote PC (even over the Internet) (stage II)<br> then transfer the call to the remote PC via VoIP if local, PSTN if not local (stage III)<br> record the call at each stage<br> <br> any help would be appreciated...<br> thanks,<br> dave cantera<br> <br> <pre class="moz-signature" cols="$mailwrapcol">-- The eyes of the Lord roam over the whole earth, to encourage those who are devoted to Him wholeheartedly. II Chronicles 16:9 NAB </pre> </body> </html>

I have an interesting task for my son's lacrosse team... it is the time-old telephone tree... I am pretty sure someone has already done this w/*, why re-invent the wheel?... a) coach calls in leaves a msg, others call in retrieve the msg b) coach calls in leaves a msg, kicks of a call to every parent plays msg c) coach calls in leaves a msg, kicks off a call to every parent, checks for

hi, I'm getting registration errors I can't debug... [Mar 23 11:07:20] NOTICE[2952]: chan_iax2.c:7344 socket_process: Registration of 'host2' rejected: 'Registration Refused' from: '10.10.10.82' I was getting a 'Cause Code: 29' INV,POKE,...,REJ but I can't duplicate that level of debugging again in the CLI> on host15 10.10.10.15

to all, I am available for work either US or Non-US for * consulting, configuring, integration with other business applications. have been working with * for about three years on and off and would like to do this full time. am available for on-site or remote project work. have 20+ years UNIX/Linux (SuSE, redhat, debian, knoppix, lamppix, slackware, cdrouter, etc) system and application

Hi, I know that Asterisk doesn't support Instant Messaging, but I'm trying to use the AGI function RECEIVE TEXT to implement a kind of IM service. I have a sip softphone that tries to send a message to an active channel and the AGI script that expect to receive the text through the STDIN. Two problems arise: First: How can I say to asterisk to get the message? (I see on CLI console that

Hello Fellow Asterisk Mailing ListMembers, When I tried to download the latest version of Asterisk this is what I get: http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed Where are all the latest Asterisk 1.4.x source files? Thanks in advance, -E

Dear all I have configure asterisk with 100 SIP PHONE ( SNOM ) but now thing is that my boss need phonebook feature find extention number by Pbook so i have read about it there is a feature in asterisk but it is with voicemail now i have IP SIP phone of SNOM so how to fine phone number by SIP phone ?? how to asterisk directory work ? Rgd satish patel

...ca82/attachment-0001.htm > > ------------------------------ > > Message: 9 > Date: Tue, 15 Jan 2008 06:22:27 +0530 > From: "Mayur" <mninama at varaha.com> > Subject: Re: [asterisk-users] Asterisk RFC2833 to SIP INFO DTMF > conversion erros. > To: <david.cantera at IBSOneCall.com>, "'Asterisk Users Mailing List - > Non-Commercial Discussion'" <asterisk-users at lists.digium.com> > Message-ID: > <mailman.10082.1200389991.10646.asterisk-users at lists.digium.com> > Content-Type: text/plain; charset="us-ascii&...

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: