[Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone

Sat Feb 5 10:05:18 MST 2005

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

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt
Waterman
Sent: Saturday, February 05, 2005 11:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Calling Asterisk Autoattendant With SIP
Phone

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:200 at 192.168.9.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.101:5060;branch=z9hG4bK7cc5dc1e
From: "Unknown" <sip:Unknown at 192.168.9.101>;tag=as63d4a421
To: <sip:200 at 192.168.9.102>
Contact: <sip:Unknown at 192.168.9.101>
Call-ID: 2c9110f8126bc7c12ea475460afb633c at 192.168.9.101
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 ----- 
From: Chamberland-Larose,  <mailto:guillaume at ea.com> Guillaume 
To: Asterisk Users Mailing List -
<mailto:asterisk-users at lists.digium.com> Non-Commercial Discussion 
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

So, I would suggest you check out the asterisk handbook here:
http://www.digium.com/handbook-draft.pdf

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

  _____  

From: Matt Waterman [mailto:waterman at dehp.net] 
Sent: Wednesday, February 02, 2005 7:08 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone

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 Asterisk at Home 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

  _____  

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