Voxalot / SIP Broker Support Forums

Voxalot / SIP Broker Support Forums (https://forum.sipbroker.com/index.php)
-   SIP Broker Support (https://forum.sipbroker.com/forumdisplay.php?f=9)
-   -   Asterisk and Sipbroker, please help (https://forum.sipbroker.com/showthread.php?t=3485)

djrh 10-25-2008 05:07 AM

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 test@sip.dyndns.org, where sip.dyndns.org is my Asterisk box. Then I add an alias say 666 to test@sip.dyndns.org (the number is *0111666). as per the FAQ section, I added the following section in sip.conf:

Code:

[general]
externip=sip.dyndns.org

[sipbroker]
type=peer
fromuser=test
fromdomain=sip.dyndns.org
host=sipbroker.com
port=5060
insecure=very
nat=yes
canreinvite=no
context=sipbroker_inbound

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:

Code:

[sipbroker_inbound]
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?

PLEASE HELP :(
Thanks in advance.

djrh 10-25-2008 06:24 PM

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:
Code:

Host                            Username      Refresh State
us.voxalot.com:5060            xxxxxx            585 Registered

and

Code:

* 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      : us.voxalot.com
  Addr->IP    : 64.34.173.199 Port 5060
  Defaddr->IP  : 0.0.0.0 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

Code:

register => xxxxxx:password:xxxxxx@us.voxalot.com/xxxxxx
and the section in sip.conf:

Code:

[voxalot]
type=friend
username=xxxxxx
secret=password
host=us.voxalot.com
insecure=very
qualify=yes
nat=yes
canreinvite=yes
context=voxalot_in

I've restarted my box several times without any luck.
PLEASE HELP :(

PS
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


All times are GMT. The time now is 12:27 AM.

Powered by vBulletin® Version 3.7.2
Copyright ©2000 - 2024, Jelsoft Enterprises Ltd.