I think you need to disable the sip alg on the sonicwall.
On Wed, Sep 8, 2010 at 10:19 AM, Stiles Watson <wat...@datatek-net.com>wrote:
> 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><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><sipx-users-boun...@list.sipfoundry.org>
> To: Discussion list for users of sipXecs software
> <sipx-users@list.sipfoundry.org> <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><tgrazi...@myitdepartment.net>
> To: "Discussion list for users of sipXecs software"
> <sipx-users@list.sipfoundry.org> <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 listsipx-us...@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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