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



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