21 May
2013
21 May
'13
2:47 p.m.
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Hi,
I've got my Kamailio script working, hurray! I had created SRV and NAPTR
records for the proxies but was missing a critical line in the script:
dns_try_naptr=yes
This enables Kamailio to query the NAPTR and SRV records and then decide
on the appropriate traffic to talk to the other proxy for non local calls.
http://kamailio.net/dokuwiki/doku.php/core-cookbook:3.0.x#dns_try_naptr
This was tough for a Kamailio noob like myself :-) I've put the whole
thing on my wiki in case anybody else would like to build a federated
SIP system.
https://www.johncahill.net/wiki/index.php/Skype_like_conferencing_System
Thanks again for the help Daniel. I will update your wiki also, (which
I shamelessly ripped off), if you give me write access.
Cheers,
John
On 21/05/13 13:15, johnc wrote:
Hi,
I've worked out what's going wrong and I fear there is a mistake in
your
config at:
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
inter-domain calls are going in plain text over UDP. I can see the
invites at the
reciving end doing a packet capture using tcpdump. My
firewall blocks these thus explaining why nothing was seen in the
kamailio log at that end.
tcpdump -vvv -i eth0 src 46.43.48.253 and udp and port 5060 >debug.txt
How do I make all inter-domain traffic only TLS?
I've ensured that there are just valid TLS SRV records at both ends:
host -t SRV _sips._tcp.unspeak.im
_sips._tcp.unspeak.im has SRV record 0 0 5061 unspeak.im.
host -t SRV _sips._tcp.atoll.dmcip.com
_sips._tcp.atoll.dmcip.com has SRV record 0 10 5061 atoll.dmcip.com.
Many thanks for all your help so far.
Regards,
John
-------- Original Message --------
Subject: Re: [SR-Users]
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
Date: Tue, 21 May 2013 05:28:31 +0100
From: johnc <johnc(a)aktivix.org>
Reply-To: Kamailio (SER) - Users Mailing List
<sr-users(a)lists.sip-router.org>
To: miconda(a)gmail.com, "Kamailio (SER) - Users
Mailing List"
<sr-users(a)lists.sip-router.org>
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Hi,
Here is a The kamailio log from the sip proxy (atoll.dmcip.com) of the
caller. There is nothing in the called server's (unspeak.im) logs from
the call.
johnc(a)atoll.dmcip.com ----Call---> gnuday(a)unspeak.im
The call gets as far as "SIP 100 trying".
Both servers have only port 5061 TCP open on the firewall to only
allow TLS.
Any help would be greatly appreciated.
Many thanks.
Regards,
John
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