Re: [asterisk-users] Cannot call to my server with SIP

2011-04-25 Thread Jamie A. Stapleton 
If you want anonymous callers to be able to place calls to Asterisk, you need 
to set allowguest=yes.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Saturday, April 23, 2011 9:40 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cannot call to my server with SIP

Op 22-04-11 23:49, Jamie A. Stapleton schreef:
 I can see your server just fine...
 
 -bash-3.2# ./svmap.py xen8.vandervlis.nl
 | SIP Device | User Agent  | Fingerprint |
 --
 | 91.198.178.28:5060 | Asterisk PBX 1.6.2.9-2+squeeze1 | disabled|
 
 However, if I try to call, Asterisk is saying:
 -- Called p...@vandervlis.nl
 [2011-04-22 17:47:13] NOTICE[10639]: chan_sip.c:19036 handle_response_invite: 
 Failed to authenticate on INVITE to ...;tag=as131f7b6a'

Ah, this is very good information. I see you, but I don't understand why
I don't see myself when I try this. Maybe my sip client (Ekiga) is not OK.

Asterisk log:
[Apr 22 23:46:50] NOTICE[29497] chan_sip.c: Sending fake auth rejection
for device Jamie A. Stapleton
sip:2233440...@sip2sip.info;tag=0wqaLzsAyMQwTdfcP2r0mG2FkPBQjEQF

Firewall log:
Apr 22 23:46:50 xen8 kernel: [3824476.043190] FW:IN=eth0 OUT=
MAC=00:16:3e:72:6f:93:00:14:f6:7e:d7:f0:08:00 SRC=81.23.228.150
DST=91.198.178.28 LEN=1320 TOS=0x08 PREC=0x00 TTL=61 ID=0 DF PROTO=UDP
SPT=5060 DPT=5060 LEN=1300
Apr 22 23:46:50 xen8 kernel: [3824476.043556] FW:IN= OUT=eth0
SRC=91.198.178.28 DST=81.23.228.150 LEN=782 TOS=0x00 PREC=0x00 TTL=64
ID=17809 PROTO=UDP SPT=5060 DPT=5060 LEN=762
Apr 22 23:46:50 xen8 kernel: [3824476.048153] FW:IN=eth0 OUT=
MAC=00:16:3e:72:6f:93:00:14:f6:7e:d7:f0:08:00 SRC=81.23.228.150
DST=91.198.178.28 LEN=411 TOS=0x08 PREC=0x00 TTL=61 ID=0 DF PROTO=UDP
SPT=5060 DPT=5060 LEN=391

 What do you have allowguest 
 (http://www.voip-info.org/wiki/view/Asterisk+sip+allowguest) set to?

I was testing security. It's like this:

sip.conf:
---
[general]
context=default
allowguest=no
alwaysauthreject=yes
(...)

[guests]
context=default
allowguest=yes

[trunk]
context=dialout
(...)

[phone-paul]
context=dialout
(...)

[phone-ann]
context=dialout
(...)
---

extensions.conf:
-
[default]
include = users

[dialout]
include = users
exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT)

[users]
exten=6001,1,Dial(SIP/paul,20)
exten=6002,1,Dial(SIP/ann,20)
(...)


Thanks for your help!

With regards,
Paul van der Vlis.




-- 
http://www.vandervlis.nl/



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Re: [asterisk-users] Cannot call to my server with SIP

2011-04-23 Thread Paul van der Vlis 
Op 22-04-11 22:58, Steve Edwards schreef:
 Op 22-04-11 18:13, Eric Wieling schreef:

 sip set debug on should help
 
 On Fri, 22 Apr 2011, Paul van der Vlis wrote:
 
 I've tried it, but no, nothing...
 
 Sounds like you have very basic network issues.
 
 Can this host ping your SIP endpoint?
 
 Can this host ping any other host on your network?
 
 Can this host ping any host out on the Internet like 8.8.8.8?

I am testing with accounts on public services (ekiga.net and
sip2sip.info). I cannot run test from this machines. But I think my
network is OK, you can try yourself to ping to my machine
xen8.vandervlis.nl.

I would like to have some test-tool

 'sudo tcpdump' and sudo tcpdump port sip' may shed some clues.

It's am idea, but letting my firewall log is something like this.

With regards,
Paul van der vlis.





-- 
http://www.vandervlis.nl/



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Re: [asterisk-users] Cannot call to my server with SIP

2011-04-23 Thread Paul van der Vlis 
Op 22-04-11 23:49, Jamie A. Stapleton schreef:
 I can see your server just fine...
 
 -bash-3.2# ./svmap.py xen8.vandervlis.nl
 | SIP Device | User Agent  | Fingerprint |
 --
 | 91.198.178.28:5060 | Asterisk PBX 1.6.2.9-2+squeeze1 | disabled|
 
 However, if I try to call, Asterisk is saying:
 -- Called p...@vandervlis.nl
 [2011-04-22 17:47:13] NOTICE[10639]: chan_sip.c:19036 handle_response_invite: 
 Failed to authenticate on INVITE to ...;tag=as131f7b6a'

Ah, this is very good information. I see you, but I don't understand why
I don't see myself when I try this. Maybe my sip client (Ekiga) is not OK.

Asterisk log:
[Apr 22 23:46:50] NOTICE[29497] chan_sip.c: Sending fake auth rejection
for device Jamie A. Stapleton
sip:2233440...@sip2sip.info;tag=0wqaLzsAyMQwTdfcP2r0mG2FkPBQjEQF

Firewall log:
Apr 22 23:46:50 xen8 kernel: [3824476.043190] FW:IN=eth0 OUT=
MAC=00:16:3e:72:6f:93:00:14:f6:7e:d7:f0:08:00 SRC=81.23.228.150
DST=91.198.178.28 LEN=1320 TOS=0x08 PREC=0x00 TTL=61 ID=0 DF PROTO=UDP
SPT=5060 DPT=5060 LEN=1300
Apr 22 23:46:50 xen8 kernel: [3824476.043556] FW:IN= OUT=eth0
SRC=91.198.178.28 DST=81.23.228.150 LEN=782 TOS=0x00 PREC=0x00 TTL=64
ID=17809 PROTO=UDP SPT=5060 DPT=5060 LEN=762
Apr 22 23:46:50 xen8 kernel: [3824476.048153] FW:IN=eth0 OUT=
MAC=00:16:3e:72:6f:93:00:14:f6:7e:d7:f0:08:00 SRC=81.23.228.150
DST=91.198.178.28 LEN=411 TOS=0x08 PREC=0x00 TTL=61 ID=0 DF PROTO=UDP
SPT=5060 DPT=5060 LEN=391

 What do you have allowguest 
 (http://www.voip-info.org/wiki/view/Asterisk+sip+allowguest) set to?

I was testing security. It's like this:

sip.conf:
---
[general]
context=default
allowguest=no
alwaysauthreject=yes
(...)

[guests]
context=default
allowguest=yes

[trunk]
context=dialout
(...)

[phone-paul]
context=dialout
(...)

[phone-ann]
context=dialout
(...)
---

extensions.conf:
-
[default]
include = users

[dialout]
include = users
exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT)

[users]
exten=6001,1,Dial(SIP/paul,20)
exten=6002,1,Dial(SIP/ann,20)
(...)


Thanks for your help!

With regards,
Paul van der Vlis.




-- 
http://www.vandervlis.nl/



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Re: [asterisk-users] Cannot call to my server with SIP

2011-04-23 Thread Eric Wieling 

If you don't see the call coming in when you have sip debug enabled, then the 
call is not making it to your server.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Friday, April 22, 2011 4:35 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cannot call to my server with SIP

Op 22-04-11 18:13, Eric Wieling schreef:

 sip set debug on should help

I've tried it, but no, nothing...

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[asterisk-users] Cannot call to my server with SIP

2011-04-22 Thread Paul van der Vlis 
Hello,

I cannot call my server over the internet with SIP anymore.

Even when I do a maximum logging on my firewall, I don't see packets
coming from outside. I've tried it from an ekiga.net account and an
sip2sip.info account. What could be wrong?  I would expect incoming
traffic on port 5060 UDP...

The account is p...@vandervlis.nl. This should connect trought DNS to
the machine xen8.vandervlis.nl:
---
paul@server2:~$ host -t SRV _sip._udp.vandervlis.nl
_sip._udp.vandervlis.nl has SRV record 0 5 5060 xen8.vandervlis.nl.
---

Is here maybe somebody with an idea, or a way to debug this?
Maybe with a nice Linux commandline tool?

With regards,
Paul van der Vlis.




-- 
http://www.vandervlis.nl/



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Re: [asterisk-users] Cannot call to my server with SIP

2011-04-22 Thread Mark Deneen 
On Fri, Apr 22, 2011 at 11:02 AM, Paul van der Vlis p...@vandervlis.nl wrote:
 Hello,

 I cannot call my server over the internet with SIP anymore.

 Even when I do a maximum logging on my firewall, I don't see packets
 coming from outside. I've tried it from an ekiga.net account and an
 sip2sip.info account. What could be wrong?  I would expect incoming
 traffic on port 5060 UDP...

 The account is p...@vandervlis.nl. This should connect trought DNS to
 the machine xen8.vandervlis.nl:

When you say that you can't call _anymore_ did it work in the past?
How long ago?

Here's what I could see from here:
* xen8.vandervlis.nl is listening on udp/5060
* there is a srv record published by the authoritative name servers at
ns1, ns2 and ns3.vandervlis.nl

The only thing that I noticed was that the TTL for xen8.vandervlis.nl
was 24 hours, which is why I asked about it working in the past.  Is
it possible that the ip address on xen8.vandervlis.nl has changed but
the old record is still present in some dns caches?

-M

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Re: [asterisk-users] Cannot call to my server with SIP

2011-04-22 Thread Eric Wieling 

sip set debug on should help

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Deneen
Sent: Friday, April 22, 2011 12:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cannot call to my server with SIP

On Fri, Apr 22, 2011 at 11:02 AM, Paul van der Vlis p...@vandervlis.nl wrote:
 Hello,

 I cannot call my server over the internet with SIP anymore.

 Even when I do a maximum logging on my firewall, I don't see packets
 coming from outside. I've tried it from an ekiga.net account and an
 sip2sip.info account. What could be wrong?  I would expect incoming
 traffic on port 5060 UDP...

 The account is p...@vandervlis.nl. This should connect trought DNS to
 the machine xen8.vandervlis.nl:

When you say that you can't call _anymore_ did it work in the past?
How long ago?

Here's what I could see from here:
* xen8.vandervlis.nl is listening on udp/5060
* there is a srv record published by the authoritative name servers at
ns1, ns2 and ns3.vandervlis.nl

The only thing that I noticed was that the TTL for xen8.vandervlis.nl
was 24 hours, which is why I asked about it working in the past.  Is
it possible that the ip address on xen8.vandervlis.nl has changed but
the old record is still present in some dns caches?

-M

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Re: [asterisk-users] Cannot call to my server with SIP

2011-04-22 Thread Paul van der Vlis 
Op 22-04-11 18:04, Mark Deneen schreef:
 On Fri, Apr 22, 2011 at 11:02 AM, Paul van der Vlis p...@vandervlis.nl 
 wrote:
 Hello,

 I cannot call my server over the internet with SIP anymore.

 Even when I do a maximum logging on my firewall, I don't see packets
 coming from outside. I've tried it from an ekiga.net account and an
 sip2sip.info account. What could be wrong?  I would expect incoming
 traffic on port 5060 UDP...

 The account is p...@vandervlis.nl. This should connect trought DNS to
 the machine xen8.vandervlis.nl:
 
 When you say that you can't call _anymore_ did it work in the past?
 How long ago?

Yes, it did work untill a few days ago.

I've changed my Asterisk configuration, but that's something else I
would say. When the configuration is wrong, I still must see traffic in
the firewall log...

 Here's what I could see from here:
 * xen8.vandervlis.nl is listening on udp/5060
 * there is a srv record published by the authoritative name servers at
 ns1, ns2 and ns3.vandervlis.nl
 
 The only thing that I noticed was that the TTL for xen8.vandervlis.nl
 was 24 hours, which is why I asked about it working in the past.  Is
 it possible that the ip address on xen8.vandervlis.nl has changed but
 the old record is still present in some dns caches?

No, the IP has not changed. So I don't understand it.

Met vriendelijke groet,
Paul van der Vlis.




-- 
http://www.vandervlis.nl/



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Re: [asterisk-users] Cannot call to my server with SIP

2011-04-22 Thread Paul van der Vlis 
Op 22-04-11 18:13, Eric Wieling schreef:
 
 sip set debug on should help

I've tried it, but no, nothing...

With regards,
Paul van der Vlis.




-- 
http://www.vandervlis.nl/



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Re: [asterisk-users] Cannot call to my server with SIP

2011-04-22 Thread Steve Edwards 

Op 22-04-11 18:13, Eric Wieling schreef:


sip set debug on should help


On Fri, 22 Apr 2011, Paul van der Vlis wrote:


I've tried it, but no, nothing...


Sounds like you have very basic network issues.

Can this host ping your SIP endpoint?

Can this host ping any other host on your network?

Can this host ping any host out on the Internet like 8.8.8.8?

'sudo tcpdump' and sudo tcpdump port sip' may shed some clues.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Cannot call to my server with SIP

2011-04-22 Thread Jamie A. Stapleton 
I can see your server just fine...

-bash-3.2# ./svmap.py xen8.vandervlis.nl
| SIP Device | User Agent  | Fingerprint |
--
| 91.198.178.28:5060 | Asterisk PBX 1.6.2.9-2+squeeze1 | disabled|

However, if I try to call, Asterisk is saying:
-- Called p...@vandervlis.nl
[2011-04-22 17:47:13] NOTICE[10639]: chan_sip.c:19036 handle_response_invite: 
Failed to authenticate on INVITE to ...;tag=as131f7b6a'

What do you have allowguest 
(http://www.voip-info.org/wiki/view/Asterisk+sip+allowguest) set to?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Friday, April 22, 2011 11:03 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Cannot call to my server with SIP

Hello,

I cannot call my server over the internet with SIP anymore.

Even when I do a maximum logging on my firewall, I don't see packets
coming from outside. I've tried it from an ekiga.net account and an
sip2sip.info account. What could be wrong?  I would expect incoming
traffic on port 5060 UDP...

The account is p...@vandervlis.nl. This should connect trought DNS to
the machine xen8.vandervlis.nl:
---
paul@server2:~$ host -t SRV _sip._udp.vandervlis.nl
_sip._udp.vandervlis.nl has SRV record 0 5 5060 xen8.vandervlis.nl.
---

Is here maybe somebody with an idea, or a way to debug this?
Maybe with a nice Linux commandline tool?

With regards,
Paul van der Vlis.




-- 
http://www.vandervlis.nl/



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