My settings for the gateway are all default - Under
"Configuration", I defined "Address" as "den.teliax.net
<http://den.teliax.net>" - Under "CallerID" I set the
"Default Caller ID" to my incoming phone number - under
"ITSP Account" I defined "Username" ("Authentication
Username" is left blank), "Password" and checked
"Register on Initialization". Everything else is defaulted.
When I do a packet capture on the WAN port of the pfSense
- I see Teliax sending me OPTION pings to the NAT'd port
number (37678 in this case). When I look at the State
table I see active states from sipX:5080 -> pfSense:37678
-> den.teliax.net:5060 <http://den.teliax.net:5060>.
Incoming Invite is to the external port (37678).
So, it looks like FreeSwitch on Teliax end is doing its
NAT compensation magic and pfSense is staying out of the
way.
Interestingly, when I looked at the packet capture and
state tables - in addition to the connection from
sipXbridge on port 5080 - there is also a connection
maintained from sipXecs on port 5060 (which in this case
is being NAT'd to port 5041). So, I am getting OPTION
pings to port 37678 (translated to 5080), to which
sipXbridge respondes "406 Not Acceptable" and OPTION
pings to port 5041 (translated to 5060) to which sipX
responses "200 Okay". The "Request URI" for the OPTION
ping to sipXbridge looks like "sip:teliaxusername@(Ext.
IP Address):37678;transport=udp;fs_nat=yes". The
"Request URI" for the OPTION ping to sipX looks like
"sip:s@(Ext IP Address):5041;fs_nat=yes".
Dave
----- Original Message -----
From: "Tony Graziano" <tgrazi...@myitdepartment.net>
<mailto:tgrazi...@myitdepartment.net>
To: sipx-users@list.sipfoundry.org
<mailto:sipx-users@list.sipfoundry.org>
Sent: Tuesday, September 7, 2010 6:20:04 PM GMT -06:00
US/Canada Central
Subject: Re: [sipx-users] Call drops after 1 min & 29 secs
Then it would be good to have a template for them. Can
you detail an example
of your gateway? Are they sending on port 5080? What did
you have to do to
get them to send on port 5080?
============================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
<mailto:tgrazi...@myitdepartment.net>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
----- Original Message -----
From: sipx-users-boun...@list.sipfoundry.org
<mailto:sipx-users-boun...@list.sipfoundry.org>
<sipx-users-boun...@list.sipfoundry.org>
<mailto:sipx-users-boun...@list.sipfoundry.org>
To: Discussion list for users of sipXecs software
<sipx-users@list.sipfoundry.org>
<mailto:sipx-users@list.sipfoundry.org>
Sent: Tue Sep 07 19:17:14 2010
Subject: Re: [sipx-users] Call drops after 1 min & 29 secs
I can report that I have 4.2.1 installed and working very
nicely with
Teliax. I have configured a gateway using very "plain
vanilla" settings and
it worked pretty much "right out of the box". Incoming
calls and outgoing.
MOH and transfers all seem to work fine. I currently have
a Grandstream
GXP-2020 and Polycom 301 on that system for
testing/evaluation and will
probably put it into "production" in the next day or two.
I have sipX
sitting behind a pfSense firewall. I am using the Denver
proxy for incoming
calls and outgoing route to their Chicago proxy.
I am limited in choices for ITSPs that can provide local
DIDs and have been
working with Teliax for about 4-5 years. I personally
find them to be pretty
good and a decent value when using the PAYG services.
Dave
----- Original Message -----
From: "Tony Graziano" <tgrazi...@myitdepartment.net>
<mailto:tgrazi...@myitdepartment.net>
To: "Discussion list for users of sipXecs software"
<sipx-users@list.sipfoundry.org>
<mailto:sipx-users@list.sipfoundry.org>
Sent: Tuesday, September 7, 2010 5:40:35 PM GMT -06:00
US/Canada Central
Subject: Re: [sipx-users] Call drops after 1 min & 29 secs
That still references using port 5060 and ip
authentication. He would need
to ensure they support using the public IP at port 5080.
It sounds like he
may have to get them to do that for him manually.
On Tue, Sep 7, 2010 at 6:29 PM, Todd Hodgen <
thod...@verizon.net <mailto:thod...@verizon.net> > wrote:
There have been some discussions about this ITSP on the
list in the past.
I did find this one.
http://forum.sipfoundry.org/index.php?t=msg&goto=44468&S=9a2fe924342a700db212b8481e97cc22#msg_44468
<http://forum.sipfoundry.org/index.php?t=msg&goto=44468&S=9a2fe924342a700db212b8481e97cc22#msg_44468>
Not sure if this fixes your problems, but it does
reference a dashboard that
you may want to access for some configuration options.
I’d search more of
the archives as well for people that have referenced this
ITSP and have
successfully gotten it working.
From: sipx-users-boun...@list.sipfoundry.org
<mailto:sipx-users-boun...@list.sipfoundry.org> [mailto:
sipx-users-boun...@list.sipfoundry.org
<mailto:sipx-users-boun...@list.sipfoundry.org> ] On
Behalf Of Tony Graziano
Sent: Tuesday, September 07, 2010 3:16 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Call drops after 1 min & 29 secs
If your firewall has a packet capture facility, you can
do a pcap on the WAN
interface and see what they are sending.
I would suspect if anyone has a working teliax config
they will share it.
On Tue, Sep 7, 2010 at 6:15 PM, Tony Graziano <
tgrazi...@myitdepartment.net
<mailto:tgrazi...@myitdepartment.net>
> wrote:
I think unless you are wed to them, it would be easier to
switch to a
"normal" provider. Supported providers in the templates
usually take 5
minutes to setup. I HOPE your firewall is doing manual
versus automatic NAT.
I looked at Teliax and they seem "residentially" focused,
and really
expensive for business plans.
On Tue, Sep 7, 2010 at 6:12 PM, Stiles Watson <
wat...@datatek-net.com <mailto:wat...@datatek-net.com> >
wrote:
Unfortunately, there is no way in the Teliax portal to
even see if you are
registered, much less what port.
The reason I had 5060 forwarded to sipx was this was how
I had Trixbox CE
setup and working. There is nothing in my Teliax setup
which I changed to
force 5060.
Thanks for the pdf. With the exception of the SIP port, I
think I have
everything setup correctly. I changed my NAT rules to
forward 5080 instead
of 5060 and the call acted exactly the same.
I've also asked Teliax if they have config info for sipX
and they said no,
but many are using the two together successfully. Here is
their exact
response:
"We do not have a have a configuration for them. However,
I know that many
customers have used SIPXECS without a problem. The main
information you need
is the username, secret, and host that you are
registering to."
I've asked them what port they are sending the INVITE on
and am waiting on a
response.
Any other suggestions/thoughts?
Stiles
Tony Graziano wrote:
It means they are not acking the call. I suspect this is
because sipxbridge
may not be involved in the call, and only sipxproxy is,
which would be
problematic for a lot of call scenarios (like transfers).
I'm confused though, because it seems you are breaking
"rule #1" when using
sipxbridge... you are having the calls sent to port 5060
instead of 5080.
When you register with teliax, can you see on their
portal what port you are
registering on? Can you confirm they are sending to you
on a specific port?
If so, what port?
You should peek at this:
http://blog.myitdepartment.net/wp-content/uploads/2009/11/Call-Setup-Example-sipXecs-through-ITSP1.pdf
Somehow I don't believe you are doing it quite like that.
On Tue, Sep 7, 2010 at 5:18 PM, Stiles Watson <
wat...@datatek-net.com <mailto:wat...@datatek-net.com> >
wrote:
Running
• sipXecs v 4.2.1
• ITSP is Teliax
• SIP ports 5060 & 5061 are routed to sipX server
• RTP ports 30000-31000 are routed to sipX server
• Polycom IP 335 hardphone
I'm able to place incoming and outgoing calls through
Teliax, but calls
consistently drop after 1 min. 29 sec.
Teliax device config change attempts:
• Enable DNIS (teliax sends number in sip invite
instead of user)
• result: calls still drop after 1 min. 29 sec.,
but made call
routing easier via a custom DID!
• Entered public IP under "Your IP"
• This is optional and resulted in not being able
to make inbound
calls (I read in the archives that this is recommended
with Teliax - is
there a sipX config change needed to make this work?)
sipX config for teliax SIP trunk Gateway:
• Configuration
• Enabled: yes • Name: teliax
• SBC Route: sipXbridge-1
• Address: den.teliax.net <http://den.teliax.net>
(this has to match with the proxy setting
in your teliax account)
• Port: 0
• Transport protocol: Auto
• Location: all
• Shared: yes
• Caller ID
• Default Caller ID: set this to the number from
Teliax •
use default for all other settings
• Dial Plan
• Enabled and added both Local & Long Distance
dial plans to this
gateway
• ITSP Account
• Username: use teliax username •
Authentication Username:
same as Username
• Password: use teliax device password
• Register on init: yes
• ITSP server address: same as Config-->Address above
• Use public address for call setup: yes (I tried
both yes and no,
calls completed either way and did not effect disconnect
problem)
• Strip private headers: default
• Use default asserted identity: default
• Asserted identity: default
• Use default preferred identity: default
• Preferred identity: default
• User part of INVITE SIP URI is a phone number: NO
• ITSP Registrar Address: default
• ITSP Registrar Port: default
• Registration interval: default
• Session Timer Interval: default
• Method to use for SIP keepalive: Empty SIP
message (also tried
None)
• Method to use for RTP keepalive: Replay last
sent packet (also
tried None)
• Route by To Header: default
Any thoughts as to why the calls would drop after 1 min.
29 sec.?
Stiles
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--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
<mailto:tgrazi...@voice.myitdepartment.net>
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
<mailto:tgrazi...@myitdepartment.net>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net
<mailto:helpd...@voice.myitdepartment.net>
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
_______________________________________________
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<mailto:sipx-users@list.sipfoundry.org> List Archive:
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--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
<mailto:tgrazi...@voice.myitdepartment.net>
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
<mailto:tgrazi...@myitdepartment.net>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net
<mailto:helpd...@voice.myitdepartment.net>
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
<mailto:tgrazi...@voice.myitdepartment.net>
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
<mailto:tgrazi...@myitdepartment.net>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net
<mailto:helpd...@voice.myitdepartment.net>
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
_______________________________________________
sipx-users mailing list
sipx-users@list.sipfoundry.org
<mailto:sipx-users@list.sipfoundry.org>
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
<mailto:tgrazi...@voice.myitdepartment.net>
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
<mailto:tgrazi...@myitdepartment.net>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net
<mailto:helpd...@voice.myitdepartment.net>
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
_______________________________________________
sipx-users mailing list
sipx-users@list.sipfoundry.org
<mailto:sipx-users@list.sipfoundry.org> List Archive:
http://list.sipfoundry.org/archive/sipx-users/
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