Re: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone

Sat, 05 Feb 2005 09:02:18 -0800

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
 
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:[EMAIL PROTECTED]
Sent: Wednesday, February 02, 2005 7:08 PM
To: asterisk-users@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 [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

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