Tony, I'm always thankful for the help I get here and I've always tried
to be verbose in describing what I'm doing and what problems I'm having.
I'll make sure I'm more so in the future.
At one time I was hearing sipx MOH (classical guitar, correct?), but I'm
not hearing it any more.
I'll get sipx to 4.4, update one of the phones, and reconfig everything
before I come back for help.
Thanks again.
Stiles
On 03/27/2012 04:45 PM, Tony Graziano wrote:
In the voip.ms <http://voip.ms> template you would put in atlanta in
two places.
beyond that, your first description was wrong on the call flow.
how sipx uses ports on itsp calls...
FROM port 5080 (sipx) to ITSP on port 5060 (ITSP) for outbound calls.
FROM port 5060 (ITSP) to sipx on port 5080 (sipx) for calls coming TO
sipx.
You are literally having a bunch of people trying to help you, but you
really need to be accurate in your descriptions, chaos ensues. Do us
all a favor and consider a backup, yum update and reboot so you have
something new-ish and describe your call flow a little more accurately.
Besides that, the little snippets from your firewall tell us little
other than what IP's are talking to what ports. So you should ensure
your sonicwall has current supportable code in it and consider sending
a siptrace. If you confirm the music on hold is being heard AND it is
sipx MOH, the problem could be any number of things, the least of
which is phone firmware.
SIPTRACE
http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
The problem is MOH and resume. Your firewall isn't giving us diddly to
go on for that. Either get a pcap at the firewall looking at the
ITSP's address or get a siptrace (preferred) so we can see what sipx
is thinking before it does it.
We've seen this done a hundred times with sonicwall, without issue,
while we suspect phone firmware and the fact you are woefully behind
(as in sipx and phone firmware is wayyyy out of date)... hard to keep
asking folks to get in wayback machine multiple times for you.
You may also consider updating ONE phone to 3.2.6 and bootrom 4.3.1
and test just it.
On Tue, Mar 27, 2012 at 4:20 PM, Stiles Watson <[email protected]
<mailto:[email protected]>> wrote:
Todd,
No, I've not changed any port info. In the SBC SIP settings
"Public Port" is empty, and "External Port" is the default of
5080. Also, in the voip.ms <http://voip.ms> gateway, the only
thing I changed from the default template is Username,
Authentication Username, Password, and ITSP server address
(changed from sip.ca2.voip.ms <http://sip.ca2.voip.ms> to
atlanta.voip.ms <http://atlanta.voip.ms>). I set the Dial Plan to
the built-in Long Distance dial plan and added the DID as an alias
to the built-in Auto Attendant.
Stiles
On 03/27/2012 04:00 PM, Todd Hodgen wrote:
Stiles, are you by chance putting port numbers in fields that are
blank? There are usually some fields that call for a port
number that are blank, as they default to 5060, sometimes people
fill those fields out. If you have changed a field from its
default to something different, you will notice a dotted line
around it to denote it is changed from the default. You would
want to remove those and keep the defaults.
*From:*[email protected]
<mailto:[email protected]>
[mailto:[email protected]] *On Behalf Of
*Tony Graziano
*Sent:* Tuesday, March 27, 2012 12:49 PM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voip.ms <http://voip.ms> config
That looks wrong. The destination port for an outbound call
should be 5060 not 5080. If you used the stock template the
firewall is he issue here.
On Mar 27, 2012 3:38 PM, "Stiles Watson" <[email protected]
<mailto:[email protected]>> wrote:
I have a Sonicwall NSA 240. I've gone back and deleted all the
firewall rules and NAT policies related to sipx and only created
new ones for 5080 UDP and 30000-31000 UDP, leaving 5060 blocked
at this point. Sip transformations is turned off and Consistent
NAT is turned on. When I make a call I see the following in the
Sonicwall's Connections Monitor:
#
Src IP
Src Port
Dst IP
Dst Port
Protocol
Src Iface
Dst Iface
Flow Type
IPS Category
Expiry (sec)
Tx Bytes
Rx Bytes
Tx Pkts
Rx Pkts
1
174.34.146.162 (atlanta.voip.ms <http://atlanta.voip.ms>)
18530
xxx.xxx.xxx.xxx (sipx server)
30000
UDP
X1
X3
N/A
29
837786
754800
4259
3774
2
174.34.146.162
5060
xxx.xxx.xxx.xxx
5080
UDP
X1
X3
SIP Control
N/A
21
6335
5586
10
12
Connection #2, the SIP Control, is the registration connection
and stays live all the time. #1 above only appears when there is
an active call.
Stiles
On 03/27/2012 12:25 PM, Gerald Drouillard wrote:
On 3/27/2012 12:03 PM, Stiles Watson wrote:
This is where one swallows one's pride.... The way I was entering
data caused the drop-down to not be displayed.
To keep this short:
1. When you first select Add new gateway>Sip Trunk, the template
drop down is not visible. I was not aware this was the case
until yesterday. I just thought it was not there.
2. The template drop-down is only displayed after you enter a
name for the gateway and then select the default SBC.
3. If you ever click the Apply button before both the name and
SBC are entered, the drop down is never displayed.
This is why I never saw the template drop-down.
Now, having said all of that, I deleted my existing voip.ms
<http://voip.ms> gateway and created a new one using the template
drop-down. However, this did not fix my problem and everything is
as it was before. I still can not retrieve a call from hold or
cancel a transfer. I have verified in my voip.ms <http://voip.ms>
account that it is registered with the public IP and port 5080.
So it looks like we are back to a firewall problem, correct?
Yes. What kind of firewall do you have?
--
Regards
--------------------------------------
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
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sip: [email protected]
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