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<font size="-1"><font face="Arial">Hello,<br>
<br>
Thanks again for all the help, and perhaps I have to excuse myself for
replying only after so much time. <br>
I made some progress: some changes in extensions.conf, and removing the
line "register => <a class="moz-txt-link-abbreviated" href="mailto:username:passwd@sip.voipbuster.com">username:passwd@sip.voipbuster.com</a>" makes the
registration timeout errors disappear. I can make outgoing calls, and
the sound is OK. However, when I call my voip-in number, I get a
message from voipbuster saying "this user is currently not online",
UNLESS I have recently made some outgoing call, by which my username
probably gets registered at voipbuster.com for some limited time.<br>
I want, of course, to be reacheable on my voip-in number permanently,
but adding the line "register => blabla" again makes the errors
reappear with which I started this topic. So what am I doing wrong now?<br>
<br>
Thanks in advance, again,<br>
Remko<br>
</font></font><br>
Steve Totaro schreef:
<blockquote cite="mid447B10F4.3020809@asteriskhelpdesk.com" type="cite">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.
<br>
<br>
Remko Muis wrote:
<br>
<blockquote type="cite">Steve,
<br>
I will try that, but now I am at my office. Can I dial some number from
the command line ;-) ?
<br>
Thanks,
<br>
Remko
<br>
<br>
<br>
----- Original Message ----- From: "Steve Totaro"
<a class="moz-txt-link-rfc2396E" href="mailto:stotaro@asteriskhelpdesk.com"><stotaro@asteriskhelpdesk.com></a>
<br>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<a class="moz-txt-link-rfc2396E" href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a>
<br>
Sent: Monday, May 29, 2006 4:39 PM
<br>
Subject: Re: [Asterisk-Users] registration at Voipbuster times out
<br>
<br>
<br>
<blockquote type="cite">If the domain resolves you are probably OK,
they just dont reply to pings.
<br>
<br>
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.
<br>
<br>
Thanks,
<br>
Steve
<br>
<br>
Remko Muis wrote:
<br>
<blockquote type="cite">Hi Steve & Attilla,
<br>
<br>
Thanks for the quick replies!!
<br>
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.
<br>
<br>
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?
<br>
<br>
I tried your second suggestion, but to no avail. My dial statements
were:
<br>
<br>
exten =>
_0[12345789]XXXXXXXX,1,Dial,SIP/voipbuster-out/0031${EXTEN:1}
<br>
exten => _0[12345789]XXXXXXXX,2,Congestion
<br>
exten => _XXXXXXX,1,Dial,SIP/voipbuster-out/0031[b]10[/b]${EXTEN}
<br>
exten => _XXXXXXX,2,Congestion
<br>
<br>
Replacing "voipbuster-out" with <a class="moz-txt-link-abbreviated" href="mailto:username:passwd@sip.voipbuster.com">username:passwd@sip.voipbuster.com</a> does
not help.
<br>
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.
<br>
<br>
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."
<br>
<br>
Being an Asterisk-newbie, I appreciate your replies, but further
suggestions even more ...
<br>
<br>
Remko
<br>
<br>
<br>
<br>
----- Original Message ----- From: "Steve Totaro"
<a class="moz-txt-link-rfc2396E" href="mailto:stotaro@asteriskhelpdesk.com"><stotaro@asteriskhelpdesk.com></a>
<br>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<a class="moz-txt-link-rfc2396E" href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a>
<br>
Sent: Monday, May 29, 2006 3:43 PM
<br>
Subject: Re: [Asterisk-Users] registration at Voipbuster times out
<br>
<br>
<br>
<blockquote type="cite">Maybe a silly question but can you ping
sip.voipbuster.com from your asterisk box?
<br>
<br>
Second question and probably the answer, what is your dial statement in
extensions.conf? Contact:<a class="moz-txt-link-rfc2396E" href="sip:s@[MYEXTERNIP]"><sip:s@[MY EXTERN IP]></a>
<br>
<br>
One way to test is to create a dial statement like this exten =
_.,1,Dial(<a class="moz-txt-link-abbreviated" href="mailto:SIP/username:password@sip.voipbuster.com/15555555555">SIP/username:password@sip.voipbuster.com/15555555555</a>)
<br>
<br>
The s in the above is suspect. Turn on SIP debugging in the asterisk
console, make a call and see whats up.
<br>
<br>
Thanks,
<br>
Steve Totaro
<br>
<br>
Remko Muis wrote:
<br>
<blockquote type="cite">Hi,
<br>
<br>
I am new here on this list, and have a problem of which I hope that
somebody here can help me with it.
<br>
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 <a class="moz-txt-link-abbreviated" href="mailto:XXXXXX@sip.voipbuster.com">XXXXXX@sip.voipbuster.com</a>
<a class="moz-txt-link-rfc2396E" href="mailto:XXXXXX@sip.voipbuster.com"><mailto:XXXXXX@sip.voipbuster.com></a> 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:
<br>
Contact: Remko <a class="moz-txt-link-rfc2396E" href="sip:XXXXXX@[INTERNIPOFX-LITE-PC]:5060"><sip:XXXXXX@[INTERN IP OF X-LITE-PC]:5060></a>
<br>
in the log of X-lite and the following line in sip debug:
<br>
Contact:<a class="moz-txt-link-rfc2396E" href="sip:s@[MYEXTERNIP]"><sip:s@[MY EXTERN IP]></a>
<br>
I don't know whether this is a significant difference.
<br>
For further info, here is my sip.conf:
<br>
bindport=5060
<br>
bindaddr=0.0.0.0
<br>
externip=EXTERNIP
<br>
localnet=192.168.1.0/255.255.255.0
<br>
srvlookup=yes
<br>
maxexpirey=180 ; Maximum length of incoming registration we allow
<br>
defaultexpirey=160 ; Default length of incoming/outgoing registration
<br>
language=nl
<br>
<br>
;register to the voipbuster service
<br>
register => <a class="moz-txt-link-abbreviated" href="mailto:XXXXXX:YYYYYY@sip.voipbuster.com">XXXXXX:YYYYYY@sip.voipbuster.com</a>
<br>
<br>
;Add an extension for our softphone
<br>
;Copy this and change 1234 into 1235 for a second softphone (etc)
<br>
[1234]
<br>
type=friend
<br>
username=1234
<br>
secret=ZZZZZZ ; this is the .password. Change this !!
<br>
callerid=Remko
<br>
notransfer=yes
<br>
insecure=very
<br>
host=dynamic
<br>
;canreinvite=no
<br>
context=default
<br>
<br>
[1235]
<br>
type=friend
<br>
username=1235
<br>
secret=ZZZZZZ; this is the .password. Change this !!
<br>
callerid=Remko
<br>
notransfer=yes
<br>
insecure=very
<br>
host=dynamic
<br>
;canreinvite=no
<br>
context=default
<br>
<br>
;Configure the incoming calls connection
<br>
[voipbuster-in]
<br>
type=user
<br>
host=sip.voipbuster.com
<br>
secret=YYYYYY
<br>
realm=voipbuster.com
<br>
fromuser=XXXXXX
<br>
fromdomain=sip.voipbuster.com
<br>
context=incoming
<br>
canreinvite=no
<br>
insecure=very
<br>
qualify=no
<br>
nat=yes
<br>
dtmfmode=inband
<br>
disallow=all
<br>
allow=alaw
<br>
allow=ulaw
<br>
call-limit=5
<br>
<br>
;Configure the outgoing calls connection
<br>
[voipbuster-out]
<br>
type=peer
<br>
host=sip.voipbuster.com
<br>
username=XXXXXX
<br>
fromuser=XXXXXX
<br>
fromdomain=sip.voipbuster.com
<br>
secret=YYYYYY
<br>
realm=voipbuster.com
<br>
call-limit=5
<br>
dtmfmode=inband
<br>
context=default
<br>
insecure=very
<br>
qualify=no
<br>
nat=yes
<br>
canreinvite=no
<br>
disallow=all
<br>
allow=alaw
<br>
allow=ulaw
<br>
I am completely at a loss, hope somebody can help me here!
<br>
<br>
Yours sincerely,
<br>
Remko
<br>
ers
<br>
<br>
</blockquote>
<br>
_______________________________________________
<br>
<br>
</blockquote>
<br>
<br>
</blockquote>
<br>
</blockquote>
<br>
</blockquote>
<br>
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