[Asterisk-Users] registration at Voipbuster times out

[Steve Totaro]

If the domain resolves you are probably OK, they just dont reply to pings.

Type "asterisk -r" then type "sip debug" and even "set verbose 15" and 
try to dial.  Post the relevant console output.  Also, disable iptables 
for testing, just to eliminate that as an issue.

Thanks,
Steve

Remko Muis wrote:
> 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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>