Yes, I do have DNIS checked on the Teliax side. No, I do not have an IP
entered.
Teliax/FreeSwitch will send the Invite to whatever port you have registered
from - assuming FreeSwitch has recognized that you are behind NAT. In my case,
at the time, this was port 37678. I have not opened any firewall ports, the
OPTION pings from Teliax/FreeSwitch maintain a connection through the firewall.
One down side to this - if you flush your firewall State tables, or they get
flushed somehow, you will not receive any incoming calls until you either place
a call or until the registration in sipX expires and it re-registers (I think
this is the case, I haven't fully tested. But when I flushed the state tables,
incoming calls were broken, but were working again this morning - so I am only
assuming that this is because sipX re-registered and the connection was
re-established through the firewall).
Dave
----- Original Message -----
From: "Stiles Watson" <wat...@datatek-net.com>
To: "Discussion list for users of sipXecs software"
<sipx-users@list.sipfoundry.org>
Sent: Wednesday, September 8, 2010 9:19:27 AM GMT -06:00 US/Canada Central
Subject: Re: [sipx-users] Call drops after 1 min & 29 secs
That was my initial sipX setup as well (except I had Auth User set equal to
User).
On the Teliax side under device settings did you do either of the following?
• enable DNIS so they send the number instead of the user in the SIP
INVITE?
• enter your pubilc IP
The reason I ask is because the " User part of INVITE SIP URI is a phone
number" checkbox under the sipX ITSP Account settings defaults to 'enabled',
but unless you enable DNIS on the Teliax side, this is not the case (unless I'm
misunderstanding the something works).
Firewall:
I'm using a Sonicwall NSA 240. I have NAT policies which forward ports UDP
5080, UDP&TCP 5060-5061 & UDP 30000-31000 untranslated to the sipX server
(we're a small shop so everything is running on one server). Are you saying
that the invite actually comes to UDP port 37678?
Stiles
Dave Redmore wrote:
My settings for the gateway are all default - Under "Configuration", I defined
"Address" as "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. 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>
To: 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
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
<sipx-users-boun...@list.sipfoundry.org>
To: Discussion list for users of sipXecs software
<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>
To: "Discussion list for users of sipXecs software"
<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 > 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
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 ] 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
> 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 >
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 >
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 (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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List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: 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 List Archive:
http://list.sipfoundry.org/archive/sipx-users/
_______________________________________________
sipx-users mailing list
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
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: 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
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: 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
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: 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 List Archive:
http://list.sipfoundry.org/archive/sipx-users/
_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/
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