search for: gulbranson

Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified

Is this supported? I have a very simple setup where I have 2 X100P cards and a TDM10B. The TDM10B is connected to a phone that has a digital answering machine built into it. If I make an inbound call on either X100P interface it gets transferred to the TDM10B interface. If I let it ring the TDM10B interface answers the call and the greeting message of the answering machine starts. Then shortly

I''m trying to store an xsl stylesheet in the database and return it to the client, but at some point in the process all the angle brackets, etc are parsed out of the xml, so I get &lt;defaults&gt; instead of <defaults>. Anyone have any pointers how I would go about turning off that behavior? -Derek

Hi, Is there some script which can be called from a * extension to playback the recent incoming callers on a particular PSTN line? In the UK 1471 is a BT number which plays back the most recent callers number, it also gives you the option to call this number back (now charging you for this service too!). Is there anything similar in asterisk-land? thanks Mike

I just received two TDM400P cards, but I'm having problems with them. The full info is at: http://pastebin.com/824079 Extra, I'm using : libpri-1.2.3 zaptel-1.2.10 On a x86 stable Gentoo box. Kernel: 2.6.17 gcc-4.1.1, glibc-2.4-r4 Is that an hardware problem? Should I try the other card? -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog:

We are developing an on-demand teleconferencing solution. We will be billing per-minute/per-user. I've successfully gotten Asterisk to write CDR data to a postgres database, but with the way I've got things setup right now the CDR does not have the dialed conference number. We need this information in order to be able to bill. As teleconferencing is the only application of the

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

I am using realtime mysql for extensions, sip, and voicemail. Outbound call routing does not really perform well in realtime extensions due to the high number of rows in the database (300k), so I can not use it. It appears with my limited knowledge that the query method is not robust enough for large databases. Given the fact that I already have realtime and mysql configured, what are my options

I'm looking for an application that can monitor a channel for voice input and then proceed on. The closest thing I've found is BackgroundDetect, which expects DTMF. Here's what I'm doing: -Call file generated which calls someone and connects them to an extension. -Extension plays stuff, etc. etc. etc (not important) With digital or VoIP termination, this works fine, because *

Hi, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in

...lfree host=voip.teliax.com auth=md5 secret=password Good Luck! ------------------------------ Message: 3 Date: Fri, 11 Feb 2005 08:44:52 +0300 (EAT) From: "Juki" <juki@one2net.co.ug> Subject: Re: [Asterisk-Users] Asterisk not acceptingmultiple SIP phone logins To: "Roger Gulbranson" <roger@gulbranson.com> Cc: asterisk-users@lists.digium.com Message-ID: <3255.81.199.88.27.1108100692.squirrel@mail.one2net.co.ug> Content-Type: text/plain;charset=iso-8859-1 Sorry, I had omitted the second SIP phone details but they do exist and are at different SIP addresses as...