[Asterisk-Users] registration at Voipbuster times out

[Josep Aguilar]

Is it possible that voipbuster refuses to connect to asterisk?, perhaps
asterisk agent is blacklistet by them
Josep

-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] En nombre de Remko Muis
Enviado el: lunes, 29 de mayo de 2006 16:51
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] registration at Voipbuster times out

Steve,
I will try that, but now I am at my office. Can I dial some number from the 
command line ;-) ?
Thanks,
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 4:39 PM
Subject: Re: [Asterisk-Users] registration at Voipbuster times out


> 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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