[Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone

Sat Feb 5 11:14:33 MST 2005

Oh wow, look at that. Don't I feel stupid, haha. All right, it works now, thanks! :/

Matt
  ----- Original Message ----- 
  From: Robert Webb 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  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

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  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, Guillaume 
    To: Asterisk Users Mailing List - 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

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      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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