Hi,
We have the following network architecture :
UAC1-kamailioVoipSwitch-PSTN--Phone1
(Sip Client)
Now UAC1 calls Phone1 and everything is ok. If UAC1 hangs up session
is terminated cleanly.
But if Phone1 hangs up the BYE message which
Hi Dave,
On Fri, May 18, 2012 at 11:27 PM, Dave Platt dpl...@radagast.org wrote:
In our app we do not forward packet immediately. After enough packet
received to increase rtp packetization time (ptime) the we forward the
message over raw socket and set dscp to be 10 so that this time
packets
Its rather surprising that i'm unable to find the code documentation
generated by make progdocs. It should be /usr/share or
/usr/local/share but it does not appear to be there.
Any clue?
--
-aft
--
_
-- Bandwidth and
I have following sip account :
Name/username HostDyn
Forcerport ACL Port Status Description
demo-alice/demo-alice 192.168.7.47 D
N 1080 Unmonitored
demo-bob/demo-bob 192.168.7.47
On Thu, May 10, 2012 at 11:50 AM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/10/2012 09:39 AM, Arif Hossain wrote:
I have following sip account :
Name/username Host Dyn
Forcerport ACL Port Status Description
demo-alice/demo
On Thu, May 10, 2012 at 5:52 PM, Kevin P. Fleming kpflem...@digium.com wrote:
You'll have to provide more details (primarily a CLI log) then in order for
anyone to be able to help you. You said that Asterisk shows extension is
rejected, but extensions don't get rejected. Extensions can be 'not
We use a obfuscation software to encrypt/mangle both SIP/RTP which sits
before asterisk. What happens is sometimes we don't get any voice. after
some rtp set debug we found out that when received ip of the rtp stream
is router's public ip, everything works cleanly. But sometimes we get the
private
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
We use a obfuscation software to encrypt/mangle both SIP/RTP which sits
before asterisk. What happens is sometimes we don't get any voice. after
some rtp set debug we found out that when received ip of the rtp stream
is router's public ip, everything
Hi,
I'm using a packet interception module for modifying udp packets coming
to asterisk sip port. now my packet modification application
successfully forwards the packet but somehow there is no response from
asterisk. it may be that the modifications destroyed sanity of the sip
packet so asterisk
