[Asterisk-Users] Seting up for afirst time -- can not call

[Race Vanderdecken]

Okay, 

About the secret, comment out the line. You do have to set the secret in
the phone. So when the INVITE is exchanged Asterisk will ask the phone
for the secret, no secret, no connection.

I don't have a polycom phone so that is about all I can help with.

Oh yeah, you need a context [from-sip]

[from-sip]
exten => 101,1,Dial,(SIP/polycom_sp300_ext101)
exten => 102,1,Dial,(SIP/polycom_sp300_ext102)

As far as I know when the calls come into asterisk via SIP asterisk
checks the [from-sip context] be default.

Remember that Asterisk is first a PBX, then a VoIP/SIP Server. SIP is
sort of step-child status in Asterisk.

Race "The Tyrant" Vanderdecken.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rudolf
Ladyzhenskii
Sent: Saturday, February 26, 2005 1:08 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Seting up for afirst time -- can not call

Hi, all

I am setting up Asterisk for the first time and have some problems.

Setup is very simple -- Astersik box and two Polycom SP300 phones. I
will 
add bells and whistles as I go, at the moment things are very simple. No

TFTP servers, so phones run with their default configuration.
I set up IP addresses, netmask and gateway IPs manually on the phones.

Now, I have read of problems with polycom phones. Here is my sip.conf
file:

; SIP configuration file

[general]
port=5060
bindaddr=0.0.0.0
context=default

[polycom_sp300_ext101]
type=user
host=192.168.1.101
secret=101
context=default

[polycom_sp300_ext101]
type=peer
secret=101
host=192.168.1.101
context=default
callerid="Ext 101"

[polycom_sp300_ext102]
type=user
host=192.168.1.102
secret=101
context=default

[polycom_sp300_ext102]
type=peer
secret=102
host=192.168.1.102
context=default
callerid="Ext 102"


First question is about the secret. Should I set up something on teh
phone? 
Is it phone password (default 456)?

Now, I am trying to have some extensions. So I have edited the 
extensions.conf file and changed the [default] section:
[default]
;
; By default we include the demo.  In a production system, you
; probably don't want to have the demo there.
;
;include => demo
exten => 101,1,Dial,(SIP/polycom_sp300_ext101)
exten => 102,1,Dial,(SIP/polycom_sp300_ext102)

The rest of the file is "as is" as it came with Asterisk.

Now I run 'reload' command as CLI.

Is ist all I have to do to be able to call between those two phones? If
I 
try to call from one phone to another, after I enter first two digits
'10', 
I get "connecting" on phone screen and instant busy tone.

Any help is greatly appreciated.

Thanks,
Rudolf 

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