Taking a look at the DEBUG statements that are associated with the
thread processing the SIP response:
[Mar 15 13:16:05] DEBUG[27947] netsock2.c: Splitting 'FQDNz:5060' into...
[Mar 15 13:16:05] DEBUG[27947] netsock2.c: ...host 'FQDNz' and port '5060'.
[Mar 15 13:16:08] DEBUG[27947] netsock2.c:
hi satish,
try to debug rtp on that ip and look rtp flow you can also test
directmedia=no i encounter this as well i server is on public ip and
clients connect via vpn , vpn server is also same asterisk server calls
come in via public ip and go to call center via vpn i solved this by
hi,
try srvlookup=yes
On Tue, Mar 19, 2013 at 3:15 AM, Jaap Winius jwin...@umrk.nl wrote:
Hi folks,
Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9
to 1.8.13, my server is no longer able to register a connection to a SIP
account at my ISP (XS4ALL in the
On Tue, 19 Mar 2013 11:58:22 +0100, Asghar Mohammad wrote:
try srvlookup=yes
Already tried that, but enabling DNS lookups makes no difference when
establishing the SIP connection. The error message that I keep seeing at
the console looks like this:
[Mar 19 12:47:21] NOTICE[7494]:
I don't believe the headsets are at fault. An agent will have a number
of calls that work just fine, then with no apparent change by the agent,
a few calls in a row will not work. In some cases, the problem seems to
correct itself. In other cases, restarting the agent's computer seems
to
Thanks for the suggestions.
1) directmedia was taking the default of yes. I set to no. Will
watch and see.
2) NAT is turned off (nat=no). I've never done any RTP debugging. Is
that rtp set debug on ip 1.2.3.4? How would I interpret the output?
3) mixmonitor recordings are stored on a
hi,
rtp set debug ip 1.2.3.4
On Tue, Mar 19, 2013 at 2:09 PM, Mitch Claborn mitch...@claborn.net wrote:
Thanks for the suggestions.
1) directmedia was taking the default of yes. I set to no. Will
watch and see.
2) NAT is turned off (nat=no). I've never done any RTP debugging. Is
that
witch softphone you are using? on client pc installed some kind of
virtualpc like vmware or virtualbox? client pc have more then one network
interfaces?
you can capture sip invites from soft phone by enabling debug on client ip
sip set debug ip ip of softphon upload sip trace then somebody can
Hi All,
Using Asterisk 1.6.0.28, having to register some Cisco 7940/60 with
SIP firmware 7.4.0. Most functions work from the phone except blind
transfer (attended transfer from phone works fine and # PBX transfer
works). Blind transfer from the phone uses SIP Refer method. I've
seen a bunch of
Firewall can cause problem on client side. Check antivirus or firewall on
agent pc
Please provide your network setup for getting better idea of problem
On Mar 19, 2013 10:10 PM, Mitch Claborn mitch...@claborn.net wrote:
rtp debug on the calls that do not work correctly shows packets from
server
We have Ubuntu 12.04 clients, using either SFLPhone or Bria 3.
There is no NAT involved in the network at all (it is disabled in sip.conf).
Here are the SIP messages capture via wireshark on the client during one
problem call. Following these SIP messages, the wireshark trace shows
only RTP
hi,
User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429)
copy from asterisk 11 rtp.conf
rtpstart=1
rtpend=2
have you changed port range? if no then
your client sending rtp to a port higher then configured in rtp port range
and asterisk ignore that port.
try to change
This was the client sending from port 39409 to server port 13429, which
is in the range. From what I read, the rtpstart and rtpend define the
range that is available for use on the server, so I'm not sure this will
apply.
But, I've set my range to 5000 - 4. I'll find out tomorrow if it
The network is all on a single LAN segment - there is no NAT involved
anywhere. Agents do not have firewall or active anti-virus. See other
posts for a SIP trace.
Mitch
On 03/19/2013 12:45 PM, Bharat Lalcheta wrote:
Firewall can cause problem on client side. Check antivirus or firewall
on
Hi,
Looking for people who would like to test there Asterisk? Student project and
I need a couple of Asterisk user to test my Test Bed and use WireShark for some
traces.
Pete--
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Good point. I changed to 1 - 4.
Mitch
On 03/19/2013 06:17 PM, Asghar Mohammad wrote:
i had this problem with a gateway witch was configured from 1000 to 3000
and some time he was using ports above 2000 and result was one way voice
rtp port range is where asterisk expect audio, you
Did u changed rtp.conf ?
port is showing 39408. Asterisk definetly drop rtp packet for this port if
not updated in rtp.conf
Regards,
Bharat Lalcheta
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