Hello,
Thanks again for all the help, and perhaps I have to excuse myself for
replying only after so much time.
I made some progress: some changes in extensions.conf, and removing the
line "register => username:[EMAIL PROTECTED]" 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.
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?
Thanks in advance, again,
Remko
Steve Totaro schreef:
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"
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
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:[EMAIL PROTECTED] 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"
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
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:[EMAIL PROTECTED] EXTERN IP]>
One way to test is to create a dial statement like this exten =
_.,1,Dial(SIP/username:[EMAIL PROTECTED]/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 [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] IP OF X-LITE-PC]:5060>
in the log of X-lite and the following line in sip debug:
Contact:<sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]
;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
_______________________________________________
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
