[Asterisk-Users] Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)

[Marty Mastera]

> 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