, SRTP still doesn't work either though. I
have no knowledge of how to move forward on this, so some pointers
would be very much appreciated.
On 06/07/11 12:11, Da Rock wrote:
I'm having trouble setting up tls/srtp secure communications on my
Asterisk server- I'm still rather new at working
doesn't work either though. I have
no knowledge of how to move forward on this, so some pointers would be
very much appreciated.
On 06/07/11 12:11, Da Rock wrote:
I'm having trouble setting up tls/srtp secure communications on my
Asterisk server- I'm still rather new at working with Asterisk
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work.
I have sent this problem to this list a couple of
: Da Rock
asterisk-us...@herveybayaustralia.com.au
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 10, 2011 05:08
Subject: [asterisk-users] Unable to make outgoing calls with Internode
Surely there must be someone here who can
In my quest to resolve the ongoing issues I have with outgoing calls...
In the SIP invite, the SDP describes the media and RTP ports. If I run
sockstat I believe I can see the ports available, but if I run tcpdump I
see no packets pass or even get blocked at the firewall. How is it
initiated
On 01/21/11 20:28, Da Rock wrote:
On 01/21/11 03:19, Tom Rymes wrote:
On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
'481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up
On 01/22/11 20:00, Da Rock wrote:
On 01/21/11 20:28, Da Rock wrote:
On 01/21/11 03:19, Tom Rymes wrote:
On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
'481cf0543743e6bb7006991d409ed3bc
On 01/22/11 22:04, Da Rock wrote:
On 01/22/11 20:00, Da Rock wrote:
On 01/21/11 20:28, Da Rock wrote:
On 01/21/11 03:19, Tom Rymes wrote:
On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
On 01/23/11 10:18, Da Rock wrote:
On 01/22/11 22:04, Da Rock wrote:
On 01/22/11 20:00, Da Rock wrote:
On 01/21/11 20:28, Da Rock wrote:
On 01/21/11 03:19, Tom Rymes wrote:
On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non
On 01/21/11 03:19, Tom Rymes wrote:
On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
'481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up.
Have you tried disallowing re
I have an updated asterisk 1.8 server running on Freebsd 8.1, and
connecting through a Freebsd 8.1 pf firewall with a dumb modem adsl
connection (in other words FreeBSD is doing all the hard work). I am
trying to connect with Internode nodephone, but they aren't really
willing to spend the
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