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I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank

...e slots? Is > the TDM card between the two X100Ps? The TDM is above the two X100P's, before it was below them. -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host --__--__-- Message: 13 Date: Tue, 13 Apr 2004 00:16:25 -0700 (PDT) From: Ron McMillin <sipnow@sbcglobal.net> Subject: Re: [Asterisk-Users] Dial Outside SIP address from AGI To: asterisk-users@lists.digium.com Reply-To: asterisk-users@lists.digium.com --0-2089280042-1081840585=:81681 Content-Type: text/plain; charset=us-ascii Thank you. This explains it. Nathaniel Powning <nat@powni...

...t; where the noise level is better suited for the developers you need to > target to actually see this message. > -- > Steven Critchfield <critch@basesys.com> > > > --__--__-- > > Message: 2 > Date: Tue, 30 Mar 2004 21:45:19 -0800 (PST) > From: Ron McMillin <sipnow@sbcglobal.net> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] DTMF Detection Problem > Reply-To: asterisk-users@lists.digium.com > > --0-1376241818-1080711919=:58147 > Content-Type: text/plain; charset=us-ascii > > Hi, > My set up is like this >...

...t; where the noise level is better suited for the developers you need to > target to actually see this message. > -- > Steven Critchfield <critch@basesys.com> > > > -- __--__-- > > Message: 2 > Date: Tue, 30 Mar 2004 21:45:19 -0800 (PST) > From: Ron McMillin <sipnow@sbcglobal.net> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] DTMF Detection Problem > Reply-To: asterisk-users@lists.digium.com > > --0-1376241818-1080711919=:58147 > Content-Type: text/plain; charset=us-ascii > > Hi, > My set up is like this >...

Hi all, I have encountered this problem: if the caller is connected to the callee using Dial() command called from extensions in extensions.conf, there is no problem. But if the same caller and callee are connected using an AGI->exec('Dial'...), the line is disconnected when asnwer. There's a problem bridging. If the codecs are the same on both ends then there is no problem.

Hi all, If I'm just using Asterisk as PBX and calls going through between ouside lines and inside extensions, (not using any softphone running on the asterisk pc), does what soundcard I use affect voice quality at all? Do I have to get a full duplex soundcard? Thanks Ron

Hi all, Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use [from-sip] exten => 7723,1,Dial(SIP/897224@fwd) and this works whereas when I'm inside agi app, $AGI->exec('Dial',"SIP/897224@fwd") and this DOESN'T work. There some errors about invalid argument. If I were to do

Hi all, Do you know if there's any free public SIP proxy server that is more reliable that FWD and Iptel? Thanks Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040415/85192715/attachment.htm