I am using asterisk at home with a Cisco ATA186 and a clone X100P card. My inbound telco line is plugged into the X100P card. My telco line is also plugged into other phones in the house for now so someone else can answer the phone without asterisk being involved. What I would like to do is if someone has answered the call on a normal phone in the house I would like to be able to join the call from a SIP phone by dialing an extension or feature code. Is there any way to do this? Thanks in advance.
In your scenario, has asterisk picked up at all? -Mark> > I am using asterisk at home with a Cisco ATA186 and a clone X100P card. > My inbound telco line is plugged into the X100P card. > My telco line is also plugged into other phones in the house for now > so someone else can answer the phone without asterisk being involved. > > What I would like to do is if someone has answered the call on a > normal phone in the house I would like to be able to join the call > from a SIP phone by dialing an extension or feature code. > > Is there any way to do this? > > Thanks in advance. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
No, Asterisk saw the call but someone has already picked it up via the telco line NOT via Asterisk. So * shows as no call connected and Zap channel not in use. On Tue, 27 Jul 2004 0:23:26 +0000, Mark Woods <asteriskadmin@fuse.net> wrote:> In your scenario, has asterisk picked up at all? > > -Mark > > > > > > > I am using asterisk at home with a Cisco ATA186 and a clone X100P card. > > My inbound telco line is plugged into the X100P card. > > My telco line is also plugged into other phones in the house for now > > so someone else can answer the phone without asterisk being involved. > > > > What I would like to do is if someone has answered the call on a > > normal phone in the house I would like to be able to join the call > > from a SIP phone by dialing an extension or feature code. > > > > Is there any way to do this? > > > > Thanks in advance. > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
I have to admit that your question interested me because I'm thinking of setting up the same thing. As of yet, though, I haven't found an answer to it. It's fairly simple when * has picked up, but I haven't really devoted much time to figuring out how to do it when it hasn't. So...let me work on it, and I'll let you know what I come up with. It's going to take at least a week, though, as I'm going to Oshkosh for the EAA Airventure on Thursday...but I'll see what I can come up with after that. -Mark> > No, Asterisk saw the call but someone has already picked it up via the > telco line NOT via Asterisk. So * shows as no call connected and Zap > channel not in use. > > On Tue, 27 Jul 2004 0:23:26 +0000, Mark Woods <asteriskadmin@fuse.net> wrote: > > In your scenario, has asterisk picked up at all? > > > > -Mark > > > > > > > > > > > > I am using asterisk at home with a Cisco ATA186 and a clone X100P card. > > > My inbound telco line is plugged into the X100P card. > > > My telco line is also plugged into other phones in the house for now > > > so someone else can answer the phone without asterisk being involved. > > > > > > What I would like to do is if someone has answered the call on a > > > normal phone in the house I would like to be able to join the call > > > from a SIP phone by dialing an extension or feature code. > > > > > > Is there any way to do this? > > > > > > Thanks in advance. > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Hmmm.... I didn't notice any noise, but I was only focusing on connectivity. -Mark> > I'm getting a ton of noise on the channel just from the * side when I > pickup my zap channel. > Otherwise it works fine, if the other person in the house hangs up the > noise goes away.. > > > On Tue, 27 Jul 2004 23:30:08 -0400, Mark Woods <asteriskadmin@fuse.net> wrote: > > > > > > Kent wrote: > > > > >On Tue, 27 Jul 2004 16:58:47 +0000, Mark Woods <asteriskadmin@fuse.net> wrote: > > > > > > > > >>I have to admit that your question interested me because I'm thinking of setting up the same thing. As of yet, though, I haven't found an answer to it. > > >> > > >>It's fairly simple when * has picked up, but I haven't really devoted much time to figuring out how to do it when it hasn't. > > >> > > >>So...let me work on it, and I'll let you know what I come up with. It's going to take at least a week, though, as I'm going to Oshkosh for the EAA Airventure on Thursday...but I'll see what I can come up with after that. > > >> > > >>-Mark > > >> > > >> > > > > > >A friend of mine who is another * user suggested using an extension > > >with an empty Dial statement to connect my sip phone to the zap > > >channel. I am going to try that tonight and see if that works. > > > > > >Let me know if you figure out anything else. > > > > > >Thanks! > > > > > > > > I actually got a chance to try it just now. Works like a champ! > > > > It has the added benefit of giving direct access, with dialtone, to the > > outside line, instead of having * dial. > > > > Here's what I put in my extensions.conf: > > > > exten => 4000,1,Dial(Zap/1/) > > exten => 4000,2,Congestion > > > > -Mark > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >