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Unread 10-25-2008, 05:07 AM   #1
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Default Asterisk and Sipbroker, please help

Hi all,

I was trying to configure my asterisk to accept calls from SIPbroker for quite a while, but I must be missing something.

Say I register with SIPbroker with URI, where is my Asterisk box. Then I add an alias say 666 to (the number is *0111666). as per the FAQ section, I added the following section in sip.conf:


Since my goal is to make SIPbroker call my telephone attached to my SPA (for which I have a sip user "spaphone"), I added the following in extensions.conf:

exten => 666,1,Answer
exten => 666,2,Dial(SIP/spaphone)
Now, I call a PSTN gateway and dial *0111666#, but nothing happens. I run Asterisk in verbose mode, and there is no output whatsoever (usually it shows when a call comes in or goes out).

I guess I do not understand the SIPbroker concept. Do I need to define this user "test" elsewhere? What else do I need to do?

Thanks in advance.
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Unread 10-25-2008, 06:24 PM   #2
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I suppose the problem is the dyndns service.

I registered a voxalot user and added it to my asterisks. Then, I configured sipbroker to my voxalot SIP URI. The first I called through the SIPbroker PSTN gateways, I was able to get the call on my SPA3100 attached phone.

HOWEVER, ever since, when I place a call, the call gets transfered to voxalot, but I get a message : "The person at extension xxxxxxx is unavailable. BEEP".

This means that voxalot does not call my box. There are no messages on my Asterisk console while the call is transfered to voxalot voicemail (running in vvvc mode).

On my Asterisk, I see:
Host                            Username       Refresh State            xxxxxx             585 Registered

* Name       : voxalot
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : voxalot_in
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  Expire       : -1
  Insecure     : port,invite
  Nat          : RFC3581
  ACL          : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID    : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : Port 5060
  Defaddr->IP  : Port 0
  Def. Username: xxxxxxx
  SIP Options  : (none)
  Codecs       : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (none)
  Status       : OK (57 ms)
  Useragent    :
  Reg. Contact :
My register command is

register =>
and the section in sip.conf:

I've restarted my box several times without any luck.

After numerous tries, I was able to make it ring one more time, but it failed right after. It only seems to work once and then not to work for hours
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