[Asterisk-Users] registration at Voipbuster times out

[Remko Muis]

Until now, I had never heard about VNC-tunnels. I will install TightVNC as 
soon as I am home. Thanks for the hint! The next thing to do then is posting 
the output of sip debug while dialling some number. I fear, however, that it 
will be terribly long, because of the frequent registration trials.

Best, 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 5:19 PM
Subject: Re: [Asterisk-Users] registration at Voipbuster times out


> No.  If you can ssh into the box you could tunnel VNC to a windows box and 
> try from a softphone there.  Thats how I do it.
>
> Remko Muis wrote:
>> 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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