Thanks Todd,
I did search the archives before sending the question to the list. There
is not much discussion.
Stiles
Todd Hodgen 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] *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)
o result: calls still drop after 1 min. 29 sec., but made
call routing easier via a custom DID!
* Entered public IP under "Your IP"
o 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
o Enabled: yes
o Name: teliax
o SBC Route: sipXbridge-1
o Address: den.teliax.net <http://den.teliax.net> (this has
to match with the proxy setting in your teliax account)
o Port: 0
o Transport protocol: Auto
o Location: all
o Shared: yes
* Caller ID
o Default Caller ID: set this to the number from Teliax
o use default for all other settings
* Dial Plan
o Enabled and added both Local & Long Distance dial plans to
this gateway
* ITSP Account
o Username: use teliax username
o Authentication Username: same as Username
o Password: use teliax device password
o Register on init: yes
o ITSP server address: same as Config-->Address above
o Use public address for call setup: yes (I tried both yes
and no, calls completed either way and did not effect
disconnect problem)
o Strip private headers: default
o Use default asserted identity: default
o Asserted identity: default
o Use default preferred identity: default
o Preferred identity: default
o User part of INVITE SIP URI is a phone number: NO
o ITSP Registrar Address: default
o ITSP Registrar Port: default
o Registration interval: default
o Session Timer Interval: default
o Method to use for SIP keepalive: Empty SIP message (also
tried None)
o Method to use for RTP keepalive: Replay last sent packet
(also tried None)
o 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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_______________________________________________
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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.
--
======================
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
List Archive: http://list.sipfoundry.org/archive/sipx-users/
_______________________________________________
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