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

[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-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Saturday, April 23, 2011 9:40 AM
To: asterisk-users at 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 paul at 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:2233440757 at 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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