[asterisk-users] Cannot call to my server with SIP
Jamie A. Stapleton
jstapleton at computer-business.com
Fri Apr 22 16:49:52 CDT 2011
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'
What do you have allowguest (http://www.voip-info.org/wiki/view/Asterisk+sip+allowguest) set to?
-----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: Friday, April 22, 2011 11:03 AM
To: asterisk-users at 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 "paul at vandervlis.nl". This should connect trought DNS to
the machine xen8.vandervlis.nl:
-------
paul at 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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