asterisk users - Mar 2005 - Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)

> 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

> Hmmm... I have this aweful feeling that I'm choosing the exact wrong
> time to ask a "newbie question" :)  Oh well, here it goes.
You're obviously literate, and you've put forth effort, so hopefully no 
one here will step on your toes.

> 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"
>
> *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"  (Hey, no self-deprecation allowed!)
> progressinband=no ;polycom's seem to have trouble with the default
> progressinband=never
You'll want to add something like this to extensions.conf

[sip]
exten => 175,1,Dial(SIP/175polycom,20)
exten => 175,2,Hangup

exten => 176,1,Dial(SIP/176polycom,20)
exten => 176,2,Hangup

This should allow your phones to dial one another.  Later you may want 
to add voicemail, then your dialplan for the sip phones would look 
something like this:
exten => 175,1,Dial(SIP/175polycom,20)
exten => 175,2,Voicemail(u175)
exten => 175,102,Voicemail(b175)
exten => 175,103,Hangup

(You'll also need to add the appropriate config to voicemail.conf)

Later, if you want to make all the lines on your Polycom phones work in 
a logical manner, you may want to disable the built-in call-waiting 
"feature."  Look up CheckGroup and SetGroup for that.  But that's
down
the road a bit.

> 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.
Extensions are what asterisk uses to allow one device to talk to 
another.  You can assign an extension to do just about anything, 
including dialing another device like a sip phone or a PSTN line.