----- Original Message -----
Sent: Saturday, February 05, 2005 12:05
PM
Subject: RE: [Asterisk-Users] Calling
Asterisk Autoattendant With SIP Phone
Matt,
I thought that DIAX was an IAX based phone not
SIP based. If this is the case then you need to be putting your configs in the
iax.conf not sip.conf file. I have several iax soft phones I have been testing
and have them registering with asterisk. If you want, I can email you the
config I have for them off-list.
Robert
Thanks for the encouraging advice. I actually
spent many hours searching for and reading through documentation about this
(on the wiki and in the handbook) and I couldn't figure out how Asterisk was
supposed to work as an SIP server.
Since I posted my original message I've made a
lot more progress (and spent considerably more than 15 minutes) but I still
have not managed to get it to work.
I have specified an SIP extension (many,
actually) in the sip.conf file but I cannot get DIAX to register with
Asterisk. I've tried changing just about every variable I can while
troubleshooting. One thing that is kind of suspect is what comes up after I
have it re-read the config files:
------
Messages-Waiting: no
Voice-Message: 0/0
to 192.168.9.102:5060
Retransmitting #5 (no NAT):
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.9.101:5060;branch=z9hG4bK7cc5dc1e
From: "Unknown"
<sip:[EMAIL PROTECTED]>;tag=as63d4a421
To: <sip:[EMAIL PROTECTED]>
Contact:
<sip:[EMAIL PROTECTED]>
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message=summary
Content-Type:
application/simple-message-summary
Content-Length: 42
------
192.168.9.101 is the Asterisk server and
192.168.9.102 is the machine I've been trying to get DIAX registered on. In
the past, I've specified the .102 address in the SIP config file for an
extension but at this point I can't think of anywhere where that IP address is
specified so this is a big mystery to me. Can anyone make sense of
it?
I have the following users in my sip_additionals
file (as generated by AMP):
[200]
username=200
type=friend
secret=test
qualify=no
port=5060
nat=never
mailbox=200
host=dynamic
dtmfmode=info
context=from-internal
canreinvite=no
callerid="test"
<200>
[222]
username=222
type=friend
secret=222
qualify=no
port=5060
nat=never
mailbox=556
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="Jack"
<222>
And I've tried making simpler a simpler one with
the bare minimum:
[111]
username=111
type=friend
secret=111
port=5060
I haven't been able to register with any of
these. I'm probably missing something really simple, I'm sure, but I haven't
been able to find it in all of the time I've spent and I imagine it would take
someone less time to point it out to me than it would to write a message
telling me how I shouldn't have posted.
Matt
----- Original Message -----
Sent: Thursday, February 03, 2005 2:28
PM
Subject: RE: [Asterisk-Users] Calling
Asterisk Autoattendant With SIP Phone
I believe the web page should be modified to include a
huge, red, bold, blinking "please read the asterisk handbook available here
and search the wiki and mail archives before you post a message to the
list". That would prevent so many questions on how and where to start when
first installing asterisk. :s
Page 56 to 61 explain in lots of detail and give a
working example of sip.conf with 1 phone and 1 voip provider. The whole
thing is good to read though so you might as well read the whole thing
(quickly) hehe.
The handbook assumes you know nothing about asterisk
and pretty much everything else. You shouldn't have to spend more than 15
minutes configuring this.
Guills
I'm trying to get into the world of
Asterisk in order to use the voicemail and autoattendat features (and more
stuff later) with a Redcom switch. But, I've only started and haven't
gotten to that yet. At this point my solitary goal is to talk to the
autoattendant via an SIP phone (SJPhone). I've spent countless hours
trying to find the documentation I need to accomplish my goals but
everything I find always assumes so much and I'm left lost. Plus I haven't
found a thing about setting up Asterisk as an SIP server.
I installed the [EMAIL PROTECTED] package, so I can
edit all the config files through HTTP and I can use AMP.
I've tried 'dialing' to the IP address
of the Asterisk machine with SJPhone but the call is rejected ("number not
available"). Now, how do I specify an extension number when I
'dial'?
Thanks for any help :/
Matt
_______________________________________________
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
_______________________________________________
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
