4 Jun
2013
4 Jun
'13
6:46 p.m.
On 30 May 2013 21:09, Michael Leuker <michael(a)leuker.me> wrote:
Thank you so much for pointing me in the right
direction! It was the
missing alias. Now the extensions are working, but there's one more
problem: When I call the PBX (echo-test) or from one extension to another,
I get a hangup because Asterisk doesn't seem to receive an ACK from the
client:
"Retransmission timeout reached on transmission
OGNkYTY1ZmJmY2VmMjQ2YmM4MWU1YWY0YjU3NjlhYjA for seqno 2 (Critical Response)"
Solution is going to lie in the kamailio logs I think:
"Retransmitting #5 (no NAT) to 198.23.139.21:5060: "
...for whatever reason, Kamailio is not replying to these.
-Barry Flanagan
I have attached the log for two local calls that hangup after about 30s and
the sequence for an incoming call from one of the
trunks that works without
problems. So you can map all the IPs:
--------
198.23.139.21 (5060): Kamailio
198.23.139.21 (5080): Asterisk
--------
192.168.178.33: Home network local IP
188.105.112.187: Home network external IP
--------
94.75.247.45: Trunk
--------
NAT was set to "No - RFC3581" for these captures, but I've tried all other
possibilities (including nathelper / rtpproxy) without success. Do you (or
anybody else) have any idea where to look in order to solve this problem?
On Thu, May 30, 2013 at 10:36 AM, Barry Flanagan <barry(a)flanagan.ie>wrote;wrote:
On 29 May 2013 19:23, Michael Leuker
<michael(a)leuker.me> wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Sure, here's the sequence for an inbound call
via the "LPhone" trunk
that was supposed to go through to extension 1001. The extension was set to
"NAT" in the FreePBX settings. Just ask if you need more background.
The Asterisk part looks fine. It is sending the call to
1001@198.23.139.21:5060 which I presume is your Kamailio instance.
for some reason Kamailio is not recognising the user. Could it be that
you do not have the ip 198.23.139.21 set up on Kamailio as an alias, or
that the user 1001 is registering to a different domain?
Kamailio would be looking in the location table for "username='1001' AND
domain='198.23.139.21'"
You should check the Kamailio logs for what is happening when Asterisk
sends it the INVITE for 1001(a)198.23.139.21
Hope this helps.
-Barry
On Wed, May 29, 2013 at 6:14 PM, Barry Flanagan
<barry(a)flanagan.ie>wrote;wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 29 May 2013 10:25, Michael Leuker
<michael(a)leuker.me> wrote:
> Thank you very much for sharing your insights, Barry! I am facing the
> same problem that Trevor described:
>
> Things are working just fine on their own, but as soon as FreePBX
> comes into play, calling extensions becomes impossible because of the
> different tables used. Removing the password from FreePBX (and setting the
> Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but
> even though the extension shows as registered in FreePBX, it always shows
> as busy:
>
> chan_sip.c:23237 handle_response_invite: Failed to authenticate on
> INVITE to '"xxxxxxxx"
<sip:xxxxxxxx@198.23.139.21>;tag=as72a4117a'
> -- SIP/1001-00000006 is circuit-busy
>
>
Can you do "sip set debug on" on Asterisk and make a call and post the
output?
-Barry
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users