[Asterisk-Users] registration at Voipbuster times out

[Steve Totaro]

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
>>>>>
>>>>
>>>> _______________________________________________
>>>>
>>>
>>>
>>
>