search for: phonnumber

...of doing was to get a regular ISDN PBX and add a second internal S0 bus which I'd connect to asterisk using chan_capi. Now to place a call through asterisk I'd first have to dial into asteriks and then have asterisk give me a dialtone and recognice dtmf. Is there any way to just prefix the phonnumber with lets say a 9 and then have the PBX transmit the rest of the number to asteriks so that I would not have to go through two dialtones? Any suggestions on what PBX to use if this is possible? It's supposed to be for a private home so I should not be some really expensive professional one......

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may help: asterisk*CLI> sip show registry Host Username

...lass=3DMsoNormal><span style=3D'color:#1F497D'>From: &quot;Joshua = Small&quot; &lt;sip:8001 at 192.168.107.4&gt;;tag=3Das3d465ba3<o:p></o:p></span></p> <p class=3DMsoNormal><span style=3D'color:#1F497D'>To: = &lt;sip:phonnumber at gw02.mytel.net.au&gt;;tag=3Das5937f41d<o:p></o:p></spa= n></p> <p class=3DMsoNormal><span style=3D'color:#1F497D'>Call-ID: 2f9f21865185cb9103ef86f438a79835 at 192.168.107.4<o:p></o:p></span></p> <p class=3DMsoNormal>...

...class=3DMsoNormal><span style=3D'color:#1F497D'>From: &quot;Joshua = Small&quot; &lt;sip:8001 at 192.168.107.4&gt;;tag=3Das3d465ba3<o:p></o:p></span></p> <p class=3DMsoNormal><span style=3D'color:#1F497D'>To: &lt;sip:phonnumber at gw02.mytel.net.au&gt;;tag=3Das5937f41d<o:p></o:p></spa= n></p> <p class=3DMsoNormal><span style=3D'color:#1F497D'>Call-ID: 2f9f21865185cb9103ef86f438a79835 at 192.168.107.4<o:p></o:p></span></p> <p class=3DMsoNormal>...