On Tue, Sep 7, 2010 at 11:53 PM, Dave Redmore <
dave.redm...@spigotsystems.com> 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).
>
This is because an outbound call to the ITSP will be "sent" to them on port
5060.

sipxbridge only "cares" that inbound calls are sent to port 5080 from the
itsp.

>
> 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
>
>
> _______________________________________________
> 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/
>
>
> _______________________________________________
> 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/
> _______________________________________________
> 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.
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