5 thoughts on - SPA504G Auto Answer

• What does your dialplan look like that makes the paging call?

  Listing from my Asterisk:

  ‘8000’ => 1. Set(SIP_CODEC=alaw)
  2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061)
  3. Hangup()

  Using the Web Interface on you SPA501, log in as admin and select the advanced view. Select the SIP tab, down the bottom of the page there is a section headed ‘Linksys Key System Parameters’.

  You will want settings much like

  Linksys Key System: yes Multicast Address: 224.168.168.168:6061
  Key System Auto Discovery: no Key System IP Address:
  Force LAN Codec: 711a
  Auto Ans GrPage On Active Call: no

  Select the User tab and check

  Auto Answer Page: yes

  If you have it all configured much like I have listed hear and it still doesn’t work then you need to check the firewall configuration on your Asterisk system and ensure it is allowing outbound Multicast traffic.

  Larry.

• Hmm, my SPA525G doesn’t auto-answer a page however my SPA92X do.

  The above is for paging, I use a macro to perform an intercom, here is what I have in my extensions.ael.

  {

  _8XXX => {
  &sip_intercom(${EXTEN:1});
  };

  };

  macro sip_intercom( extension ) {
  ChanIsAvail(SIP/${LOCAL(extension)},s);
  NoOp(**** Status : ${AVAILSTATUS} ****);
  switch(${AVAILSTATUS}) {
  case 1:
  Set(TIMEOUT(absolute)20);
  SIPAddHeader(Alert-Info: Ring Answer);
  SIPAddHeader(Alert-Info: Info=Alert-Autoanswer);
  SIPAddHeader(Alert-Info: info=ringAutoAnswer);
  SIPAddHeader(Call-Info:\;Answer-After=0);
  SIPAddHeader(P-Auto-Answer: normal);
  SIPAddHeader(Answer-Mode: Auto);
  Dial(SIP/${LOCAL(extension)});
  Hangup();
  break;
  case 2:
  Busy();
  Hangup();
  break;
  case 5:
  Congestion();
  Hangup();
  break;
  default:
  PlayBack(invalid);
  Hangup();
  break;
  };
  return;
  };

  Larry.

• 
  
  

  Just upgraded the firmware on my SPA525G from 7.5.4 to 7.5.6 and the paging function is now working!

• In the process of setting up another system, there is an additional requirement for the multicast paging to work.

  Asterisk will need to know where to route the multicast traffic, on your Asterisk system, check your routing table and see if there is a route to the multicast address through the interface which connects to your phones. If not, create a routing entry, in this case to 224.168.168.168
  through the desired interface.

  Larry.

• For the benefit of others I encountered a situation where I was getting one-way audio in a call regardless of it not being a paging call, this was because the negotiated codecs for the call was one other than the one selected in the ‘Force LAN Codec:’ setting.

  It would appear setting the ‘Force LAN Codec:’ to either G711u or G711a
  _always_ enforces the phone to use this codec for its Encoder regardless of what is negotiated in SIP.

  My advice, leave the ‘Force LAN Codec:’ setting at its default value which is ‘none’.

  Larry.