[Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?

[Don Fanning]

Done
---

*CLI>
*CLI>
*CLI>     
    -- Executing SetCallerID("SIP/100-1ba9", ""xxxxx"") in new stack
    -- Executing Dial("SIP/100-1ba9",
"IAX2/xxxxx at voipbuster/0015163011118") in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass:
NEW
   Timestamp: 00008ms  SCall: 00010  DCall: 00000 [213.61.187.157:4569]
   VERSION         : 2
   CALLED NUMBER   : 0015163011118
   CALLING NAME    : xxxxx
   LANGUAGE        : en
   USERNAME        : xxxxx
   FORMAT          : 2
   CAPABILITY      : 63490
   ADSICPE         : 2
   DATE TIME       : 185630951

    -- Called jfalcon at voipbuster/0015163011118
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass:
ACK
   Timestamp: 00008ms  SCall: 00306  DCall: 00010 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass:
AUTHREQ
   Timestamp: 00001ms  SCall: 00306  DCall: 00010 [213.61.187.157:4569]
   AUTHMETHODS     : 3
   CHALLENGE       : 229696652
   USERNAME        : xxxxx

Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX     Subclass:
AUTHREP
   Timestamp: 00180ms  SCall: 00010  DCall: 00306 [213.61.187.157:4569]
   MD5 RESULT      : 8b729ab88c50ba655fef99ef151ad228

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass:
ACK
   Timestamp: 00180ms  SCall: 00306  DCall: 00010 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass:
ACCEPT
   Timestamp: 00171ms  SCall: 00306  DCall: 00010 [213.61.187.157:4569]
   FORMAT          : 2

    -- Call accepted by 213.61.187.157 (format gsm)
    -- Format for call is gsm
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass:
ACK
   Timestamp: 00171ms  SCall: 00010  DCall: 00306 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass:
LAGRQ
   Timestamp: 10009ms  SCall: 00010  DCall: 00306 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass:
LAGRQ
   Timestamp: 10051ms  SCall: 00306  DCall: 00010 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 003 Type: IAX     Subclass:
LAGRP
   Timestamp: 10051ms  SCall: 00010  DCall: 00306 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX     Subclass:
LAGRP
   Timestamp: 10009ms  SCall: 00306  DCall: 00010 [213.61.187.157:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX     Subclass:
ACK
   Timestamp: 10009ms  SCall: 00010  DCall: 00306 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX     Subclass:
ACK
   Timestamp: 10051ms  SCall: 00306  DCall: 00010 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX     Subclass:
HANGUP
   Timestamp: 10934ms  SCall: 00306  DCall: 00010 [213.61.187.157:4569]
   Unknown IE 042  : Present

Ignoring unknown information element 'Unknown IE' (42) of length 1
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX     Subclass:
ACK
   Timestamp: 10934ms  SCall: 00010  DCall: 00306 [213.61.187.157:4569]
    -- Hungup 'IAX2/voipbuster/10'
  == No one is available to answer at this time
    -- Executing NoOp("SIP/100-1ba9", "DIALSTATUS=NOANSWER") in new
stack
    -- Executing NoOp("SIP/100-1ba9", "HANGUPCAUSE=0") in new stack
    -- Executing Congestion("SIP/100-1ba9", "") in new stack
  == Spawn extension (internalselections, 90015163011118, 6) exited
non-zero on 'SIP/100-1ba9' 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
Wieling aka ManxPower
Sent: Tuesday, August 16, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX
connections?

Don Fanning wrote:
> Taking in everyone's suggestions (added a username line also) here is 
> what I got.
> Still no joy
> ---
> 
> *CLI>
> *CLI>
> *CLI>
>     -- Executing SetCallerID("SIP/100-b225", ""xxxx"") in new stack
>     -- Executing Dial("SIP/100-b225",
> "IAX2/xxxxx at voipbuster/0015163011118") in new stack
>     -- Called xxxxx at voipbuster/0015163011118
>     -- Hungup 'IAX2/voipbuster/6'
>   == No one is available to answer at this time
>     -- Executing Congestion("SIP/100-b225", "") in new stack
>   == Spawn extension (internalselections, 90015163011118, 3) exited 
> non-zero on 'SIP/100-b225'

Put a Noop(HANGUPCAUSE=${HANGUPCAUSE}) and a
Noop(DIALSTATUS=${DIALSTATUS}) as the two priorities after your Dial to
see WHY the call was hungup.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120

r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. "r" makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.

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