I was under the impression that "ITSP Registrar Port" will be the port
where the ITSP is expecting to receive REGISTERs and not where the
bridge will be sending them from. Can you verify?
On Wednesday, 22 September, 2010 09:06 AM, Tony Graziano wrote:
OK. The "registration status" page at the ITSP will show you what port
you are registered on. I think that would be a good enhancement for
sipxconfig (to show the registration port), but editing the gateway
itsp account to reflect "ITSP Registrar Port" and manually set it to
5060 should show you after re-initialization at the voip.ms
<http://voip.ms> portal that the registration is at port 5060, at
which point your test can proceed.
Logically, an ITSP that requires registration will send the calls to
the same port you are registered on.
On Tue, Sep 21, 2010 at 6:25 PM, Joegen Baclor <[email protected]
<mailto:[email protected]>> wrote:
Hi Everyone,
I had a chance to test this patch using an ITSP. I used voip.ms
<http://voip.ms> to test. The call worked with bidirectional
audio and I was satisfied. The logs however shows that voip.ms
<http://voip.ms> did not send the call to port 5060 but via the
registration port 5080. Can you point me to an ITSP that insists
on sending to 5060? Or better yet, if you have an account with an
ITSP that behaves this way, would you be able to throw my test
server an inbound call via sip-trunk reg? Any help would be
appreciated. Thanks.
Joegen
On Monday, 20 September, 2010 05:49 PM, Tony Graziano wrote:
Looking at the sipxbridge log, I see the by from 201, then the
errors start. I think the question is where the bye should be
sent and ack'd. Without seeing the sipxproxy.log its kind of hard
to say. The error implies a listening error, but that is a bit of
a long message and can simply be a result of "not knowing" what
to do... the twinkle log looks plain, it shows sending the bye to
bridge on port 5080. Is sipxbridge still listening locally on
port 5080 in this environment?I just don;t know how to read it
because the sipxbridge log shows the BYE on port 5060 and the
twinkle log shows 5080.
nBYE sip:[email protected]
<mailto:sip%[email protected]>;x-sipX-nonat SIP/2.0\r\nVia:
SIP/2.0/UDP 112.201.137.176:5080 <http://112.201.137.176:5080>;
Can you explain how the call flow for a bye should work (which
service/port) and where it should be sent (directly to sipxbridge
is my guess from the client and vice versa)?
On Mon, Sep 20, 2010 at 2:19 AM, Joegen Baclor <[email protected]
<mailto:[email protected]>> wrote:
Hi Folks,
For those of you who are following the development on this
thread, I have attached a new set of twinkle log that
demonstrates a complete call that passes through 5060 coming
from a dummy ITSP. The previous log I have sent contained a
glitch that is now corrected. I needed to modify contact
creation in sipXbridge a bit so that it sends the internal IP
address when talking to the proxy. This glitch is now
corrected.
However, I am now facing a new issue. When the BYE is coming
from the called extension, sipXproxy sends a 407 for the BYE
and sipXbridge suddenly barfs an exception
"2010-09-20T05:48:59.328000Z":1188:sipxbridge:ERR:c2.ossapp.com:Thread-88:00000000:SipListenerImpl:"Unexpected
error processing response >>>> SIP/2.0 407 Proxy
Authentication Required\r\nFrom:
<sip:[email protected]>;tag=784036913\r\nTo: \"Joegen
Baclor\" <sip:[email protected]>;tag=bjome\r\nCall-ID:
kteensdeyxos...@bravia-c4\r\ncseq: 1 BYE\r\nVia: SIP/2.0/UDP
112.201.137.176:5080;branch=z9hG4bK5fe25839220b0a96ff883192d4d6e60a373835;received=192.168.1.11;rport=5080\r\nProxy-Authenticate:
Digest realm=\"c2.ossapp.com
<http://c2.ossapp.com>\",nonce=\"e669226f7847e446773d4cceeddd161a4c96f5cb\",qop=\"auth\"\r\nServer:
sipXecs/4.3.0 sipXecs/sipXproxy (Linux)\r\nDate: Mon, 20 Sep
2010 05:48:59 GMT\r\nContent-Length: 0\r\n\r\n"
javax.sip.SipException: Unexpected exception
Newbie Questions:
1. Is the proxy suppose to authenticate mid-dialog requests
from the bridge? Is this how it behaves currently?
2. What could be causing the bridge to barf? Isn't it
suppose to just relay the response to the callee since it
would know how to construct the authentication? Or is this
something I have introduced by messing around with contact?
Joegen
On Thursday, 16 September, 2010 11:51 PM, Tony Graziano wrote:
I feel a little left out because they won't approve my
openscs registration request.
On Thu, Sep 16, 2010 at 11:35 AM, Matt White
<[email protected] <mailto:[email protected]>>
wrote:
Sweet!....thats what I was hoping to hear.
I didn't think Avaya has released any code for it's new
openscs project as the website still has nothing new
from June. But was wondering if I missed something if
Avaya had actually released openscs code.
I do think its funny the openscs webpage notes that
"/*As of July Avaya no longer participates in
SIPFoundry. SIPFoundry has forked the code base and is
being maintained by a new startup company.*/"
Rather than Avaya being the one that forked it into a
new openscs project ;-)
-M
>>> Joegen Baclor 09/16/10 10:00 AM >>>
Hi Matt,
I've heard that Avaya is trying to solve this as well.
But this one is completely community/ezuce code.
Joegen
On Thursday, 16 September, 2010 09:04 PM, Matt White wrote:
Great news. This will go a long way towards increased
interop.
Out of curiosity, is any of this based on the work that
avaya was doing before the fork? Or is this 100%
community/ezuce code?
-M
>>> Joegen Baclor <[email protected]>
<mailto:[email protected]> 09/16/10 4:37 AM >>>
Hi Folks,
I just thought you'd be interested in knowing that I
have already
successfully sent a call to port 5060 and forwarded to
the bridge by
redirection. See attached log from twinkle.
Unfortutely, I am behind
a firewall controlled by the ITSP so this is not yet
tested in an actual
environment. If you take a look at the log I have
attached, the 200 OK
is now coming from sipXbridge event if the call passed
through the main
sipXproxy listener. The ACK in this case will be
misrouted because
sipXbridge is sending the external IP as its contact.
This however
should not be an issue if the actual test call came
from an entity
outside my firewall. Hopefully we will have some more
good news in the
days to come.
Joegen
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======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: [email protected]
<mailto:[email protected]>
Fax: 434.984.8431
Email: [email protected]
<mailto:[email protected]>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: [email protected]
<mailto:[email protected]>
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-dev mailing list
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--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: [email protected]
<mailto:[email protected]>
Fax: 434.984.8431
Email: [email protected]
<mailto:[email protected]>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: [email protected]
<mailto:[email protected]>
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-dev mailing list
[email protected] <mailto:[email protected]>
List Archive: http://list.sipfoundry.org/archive/sipx-dev/
--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: [email protected]
<mailto:[email protected]>
Fax: 434.984.8431
Email: [email protected] <mailto:[email protected]>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: [email protected]
<mailto:[email protected]>
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-dev mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-dev/
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