[Asterisk-Users] registration at Voipbuster times out

[Remko Muis]

Hi Steve & Attilla,

Thanks for the quick replies!!
Attilla: your suggestion sounds promising, since I know my system clock is 
not too accurate. But that is the reason I use the network time protocol 
daemon. Time and date settings are now correct.

Steve: your question about pinging the sip-proxy servers hits the nail on 
its head: I can't, even though the names resolve to ip-addresses, and I can 
ping lots of other machines in the outside world. But why?

I tried your second suggestion, but to no avail. My dial statements were:

exten => _0[12345789]XXXXXXXX,1,Dial,SIP/voipbuster-out/0031${EXTEN:1}
exten => _0[12345789]XXXXXXXX,2,Congestion
exten => _XXXXXXX,1,Dial,SIP/voipbuster-out/0031[b]10[/b]${EXTEN}
exten => _XXXXXXX,2,Congestion

Replacing "voipbuster-out" with username:passwd at sip.voipbuster.com does not 
help.
However, I did not really expect so, since the registration timeout errors 
occur while Asterisk executes chan_sip.c. I would think that registration 
fails independently of any wrong settings in extensions.conf.

Anyway, the s in the Contact-line does look suspect to me, since I have a 
voip-in number for Voipbuster, and I read on the voip-info pages that "the s 
extension is is used when there is no known called number in the context 
used."

Being an Asterisk-newbie, I appreciate your replies, but further suggestions 
even more ...

Remko



----- Original Message ----- 
From: "Steve Totaro" <stotaro at asteriskhelpdesk.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Monday, May 29, 2006 3:43 PM
Subject: Re: [Asterisk-Users] registration at Voipbuster times out


> Maybe a silly question but can you ping sip.voipbuster.com from your 
> asterisk box?
>
> Second question and probably the answer, what is your dial statement in 
> extensions.conf? Contact:<sip:s@[MY EXTERN IP]>
>
> One way to test is to create a dial statement like this exten = 
> _.,1,Dial(SIP/username:password at sip.voipbuster.com/15555555555)
>
> The s in the above is suspect.  Turn on SIP debugging in the asterisk 
> console, make a call and see whats up.
>
> Thanks,
> Steve Totaro
>
> Remko Muis wrote:
>> Hi,
>>
>> I am new here on this list, and have a problem of which I hope that 
>> somebody here can help me with it.
>> I have a Voipbuster account, with which I would like to make phone calls 
>> via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, 
>> everything is OK, but if I let Asterisk register there, it says 
>> "registration for XXXXXX at sip.voipbuster.com 
>> <mailto:XXXXXX at sip.voipbuster.com> timed out, trying again", even though 
>> all settings are precisely as in X-Lite (username, password, and 
>> sip-proxy settings). Also I am sure the right ports are forwarded or 
>> open, both in my router and in iptables (firewall of Asterisk server). 
>> The log files of X-Lite and the output of "sip debug" show no 
>> differences, except this one:
>>  Contact: Remko <sip:XXXXXX@[INTERN IP OF X-LITE-PC]:5060>
>>  in the log of X-lite and the following line in sip debug:
>>  Contact:<sip:s@[MY EXTERN IP]>
>>  I don't know whether this is a significant difference.
>> For further info, here is my sip.conf:
>>  bindport=5060
>> bindaddr=0.0.0.0
>> externip=EXTERNIP
>> localnet=192.168.1.0/255.255.255.0
>> srvlookup=yes
>> maxexpirey=180 ; Maximum length of incoming registration we allow
>> defaultexpirey=160 ; Default length of incoming/outgoing registration
>> language=nl
>>
>> ;register to the voipbuster service
>> register => XXXXXX:YYYYYY at sip.voipbuster.com
>>
>> ;Add an extension for our softphone
>> ;Copy this and change 1234 into 1235 for a second softphone (etc)
>> [1234]
>> type=friend
>> username=1234
>> secret=ZZZZZZ ; this is the .password. Change this !!
>> callerid=Remko
>> notransfer=yes
>> insecure=very
>> host=dynamic
>> ;canreinvite=no
>> context=default
>>
>> [1235]
>> type=friend
>> username=1235
>> secret=ZZZZZZ; this is the .password. Change this !!
>> callerid=Remko
>> notransfer=yes
>> insecure=very
>> host=dynamic
>> ;canreinvite=no
>> context=default
>>
>> ;Configure the incoming calls connection
>> [voipbuster-in]
>> type=user
>> host=sip.voipbuster.com
>> secret=YYYYYY
>> realm=voipbuster.com
>> fromuser=XXXXXX
>> fromdomain=sip.voipbuster.com
>> context=incoming
>> canreinvite=no
>> insecure=very
>> qualify=no
>> nat=yes
>> dtmfmode=inband
>> disallow=all
>> allow=alaw
>> allow=ulaw
>> call-limit=5
>>
>> ;Configure the outgoing calls connection
>> [voipbuster-out]
>> type=peer
>> host=sip.voipbuster.com
>> username=XXXXXX
>> fromuser=XXXXXX
>> fromdomain=sip.voipbuster.com
>> secret=YYYYYY
>> realm=voipbuster.com
>> call-limit=5
>> dtmfmode=inband
>> context=default
>> insecure=very
>> qualify=no
>> nat=yes
>> canreinvite=no
>> disallow=all
>> allow=alaw
>> allow=ulaw
>> I am completely at a loss, hope somebody can help me here!
>>
>> Yours sincerely,
>> Remko
>> ers
>>
>
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