> Hmmm... I have this aweful feeling that I'm choosing the > exact wrong time to ask a "newbie question" :) Oh well, here > it goes. > > The quick question is : "How do I dial an extension?" > (answer is probably - "you don't" in which case:) "How do I > dial my asterisk box?" - I have no outside line, I just want > to start testing things like voicemail internally. > > The details: I am not connected to the outside world yet, I > have a couple of phones in-house and I'm trying to set up an > Asterisk internal office phone network just to get my head > wrapped around the system. I have > - my linux box set up > - the phones ftp'ing their latest firmware and config files > - I can call one phone from the other using the IP address > (no asterisk > required) > - I have installed zaptel, libpri, asterisk, asterisk samples > - I have added my 2 phones to the sip.conf file (see below) > - I see the two phones if I do a "sip show peers" with the > correct IP addresses > - I've tried to set up the phones as described at > "http://www.csh.rit.edu/~adamf/IP500.html"; > > In the QuickStart guide it says that the way to test things > are working is to call extension 1000 to get an automated > message. Clearly the phones can talk to each other, I just > want to take the next step to see if they can talk to > Asterisk. Yet I can find nothing about extensions in any of > the Polycom documentation, phone buttons and menus, etc, and > I am beginning to think that the concept of an "extension" is > an analogue phone thing and just doesn't make sense for IP phones. > > Anyway, I would really appreciate someone stopping on the > shoulder, here, and helping me drag myself out of the ditch > so I can careen down the highway, obstructing other people's > progress as a newbie should... > any help would be much appreciated. I feel like I am > suffering from a fundamental disconnect. I can read and > somewhat understand the details of the documentation > regarding dialplan etc, I just don't know where the "on > ramp" is, i.e. how to even talk to Asterisk with a phone, > with my current set up. > > The only modifications I did were to added my asterisk server > IP into the sip.cfg for the Polycom ftp account and to add > the below into my /etc/asterisk/sip.conf file. Aside from > that I'm working with a "straight out of the box" asterisk > "make; make install; make samples". > > Thanks in advance, > Don > > *CLI> sip show peers > Name/username Host Dyn Nat ACL Mask > Port > Status > 176polycom 192.168.0.176 255.255.255.255 > 5060 > Unmonitored > 175polycom 192.168.0.175 255.255.255.255 > 5060 > Unmonitored > > > Added to sip.conf: > > [175polycom] > type=friend > host=192.168.0.175 > defaultip=192.168.0.175 > dtmfmode=inband > mailbox=175 > context=sip > callerid="I am Don" > progressinband=no ;polycom's seem to have trouble with the > default progressinband=never > > [176polycom] > type=friend > host=192.168.0.176 > defaultip=192.168.0.176 > dtmfmode=inband > mailbox=176 > context=sip > callerid="I am a jerk" > progressinband=no ;polycom's seem to have trouble with the > default progressinband=never >
You're almost there...you have the phones set to the 'sip' context. Edit your extensions.conf file, create a new context called [sip] (if it isn't already there), and add a dial statement to reach each phone: [sip] exten => 175,Dial(SIP/175polycom) Exten => 176,Dial(SIP/176polycom) (There are plenty of dial modifiers you can use, but there's a basic wau to get started...now just dial 175 and 176 from each phone respecitively to reach the opposite... Marty _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
