If they are expecting ten digits, this means your gateway and dialing
plans need to drop "+1" and "+" from them. The "+" is a gateway level
setting. The "1" is a dialing plan setting. The guide was just that, a
guide. This would show you how to modify it to fit your own needs.
On Sat, Nov 21, 2009 at 12:53 AM, mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com> <mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>> wrote:
I read your guide, and I don't see anything that sticks out at me
that I'm doing wrong. Verizon says they are sending 10 and
expecting 10 digits. I have enabled and tweaked the default
dialing plans. I'm using Cisco 7940s, because I had a few laying
around. I am home now, so I just tried Xlite and get the same
results. I'm getting some demo Polycom Soundpoint 450s and 550s on
Monday so I will see if they have any different results. I will go
through it again from top to bottom and make sure am not missing
anything obvious. If I still get the same results, I will send a
complete trace.
Thanks again for your help.
Tony Graziano wrote:
But there is a simpler way to do the dial plan first. Ask
Verizon what format your numbers should be. I would assume
they would accept the full "+12025551212" (all +1 dialing). If
so, the guide I have for bandwidth.com <http://bandwidth.com>
<http://bandwidth.com> has all the rules laid out. It should
work the same, only the gateway name would be different.
On Fri, Nov 20, 2009 at 6:31 PM, Tony Graziano
<tgrazi...@myitdepartment.net
<mailto:tgrazi...@myitdepartment.net>
<mailto:tgrazi...@myitdepartment.net
<mailto:tgrazi...@myitdepartment.net>>> wrote:
I see two things that make me go hmmm....
"2009-11-20T23:11:36.845000Z":
19:OUTGOING:INFO:nshpbx1.sipx.voip:Thread-17:00000000:sipXbridge:"Sent
SIP Message :\n----Remote Host:10.87.20.5---- Port:
38526----\nSIP/2.0
403 Forbidden\r\nVia: SIP/2.0/TCP
10.87.20.5;branch=z9hG4bK-sipXecs-0016a7a00a17b94001516f24875b44f09653;rport=38526\r\nVia:
SIP/2.0/TCP
10.87.20.5;branch=z9hG4bK-sipXecs-0013a7669e28a72d44481cd5a375af980b06~f493f0f98bcc796b31453652fad2d124\r\nVia:
SIP/2.0/UDP 10.87.20.254:5060;branch=z9hG4bK0ee159ab\r\nFrom:
\"Kitchin
Matthew\"
<sip:1...@sipx.voip>;tag=00127f98eaa9004405071fb9-0d560bee\r\nTo:
<sip:916155008...@sipx.voip>\r\nCall-ID:
00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254
<mailto:00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254>
<mailto:00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254
<mailto:00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254>>\r\nCSeq:
102
INVITE\r\nServer: sipXecs/4.0.4 sipXecs/sipxbridge
(Linux)\r\nContact:
<sip:~~id~bri...@10.87.20.5:5090
<http://bri...@10.87.20.5:5090>
<http://bri...@10.87.20.5:5090/
<http://10.87.20.5:5090/>>>\r\nSupported:
replaces,100rel\r\nReason:
~~id~bridge;cause=213;text=\"Relayed Error
Response\"\r\nContent-Length:
0\r\n\r\n--------------------END--------------------\n"
403 Forbidden - I'm assuming 10.87.20.5 is your sipx system? If
so, are you *sure* Verizon will accept calls from this IP?
<sip:~~id~bri...@10.87.20.5:5090
<http://bri...@10.87.20.5:5090>
<http://bri...@10.87.20.5:5090/
<http://10.87.20.5:5090/>>>\r\nSupported:
replaces,100rel\r\nReason: ~~id~bridge;cause=213;text=\"
Relayed Error
Response\"\r\nContent-Length: - This might indicate an
issue with
your UA. What is your phone you are sending this call from? The
log is too short to be helpful because it's not providing this
information. I see an open issue (XX-5823
<http://track.sipfoundry.org/browse/XX-5823>) related to
handling
the response, no action was taken because the phone used
was not
very firendly anyway.
A complete trace of a failed call would be helpful.
Directions are here:
http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer
Since this is not in production, I would remove all the
logs after
putting the log modes in the correct level, place a test
call and
do the merge. Attached the merged.xml file to an email and
send it
in a reply to the list.
You can also do the sipviewer and view it on a winpc at
your end
with no issue.
On Fri, Nov 20, 2009 at 6:17 PM, mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>
<mailto:mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>> <mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>
<mailto:mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>>> wrote:
Not sure if there was a delay of some sort, but I got
lots of
activity
now. It is below. I will dig through them and see what
I can
find. They
announced the routing update (I'm not the network guy)
and the
equipment
did learn it. I can definitely ping their .1 address.
"2009-11-20T23:11:36.831000Z":18:OUTGOING:INFO:nshpbx1.sipx.voip:Thread-17:00000000:sipXbridge:"Sent
SIP Message :\n----Remote Host:172.30.209.62---- Port:
5070----\nACK
sip:16155008...@pcelbcn0001.dsi.globalipcom.com
<mailto:sip%3a16155008...@pcelbcn0001.dsi.globalipcom.com>
<mailto:sip%3a16155008...@pcelbcn0001.dsi.globalipcom.com
<mailto:sip%253a16155008...@pcelbcn0001.dsi.globalipcom.com>>;user=phone
SIP/2.0\r\nCall-ID:
00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254.0\r\nmax-forwards:
70\r\nFrom: \"Kitchin Matthew\"
<sip:1...@pcelbcn0001.dsi.globalipcom.com
<mailto:sip%3a1...@pcelbcn0001.dsi.globalipcom.com>
<mailto:sip%3a1...@pcelbcn0001.dsi.globalipcom.com
<mailto:sip%253a1...@pcelbcn0001.dsi.globalipcom.com>>>;tag=8483786813757111981\r\nTo:
<sip:16155008...@pcelbcn0001.dsi.globalipcom.com
<mailto:sip%3a16155008...@pcelbcn0001.dsi.globalipcom.com>
<mailto:sip%3a16155008...@pcelbcn0001.dsi.globalipcom.com
<mailto:sip%253a16155008...@pcelbcn0001.dsi.globalipcom.com>>;user=phone>;tag=aprqngfrt-od3bjv2000020\r\nVia:
SIP/2.0/UDP
pcelbcn0001.dsi.globalipcom.com:5080;branch=z9hG4bKf79a1b7109513b78e1683a1505f003513933\r\nCSeq:
1 ACK\r\nRoute:
<sip:172.30.209.62:5070;transport=udp;lr>\r\nUser-Agent:
sipXecs/4.0.4 sipXecs/sipxbridge (Linux)\r\nContent-Length:
0\r\n\r\n--------------------END--------------------\n"
"2009-11-20T23:11:36.845000Z":19:OUTGOING:INFO:nshpbx1.sipx.voip:Thread-17:00000000:sipXbridge:"Sent
SIP Message :\n----Remote Host:10.87.20.5---- Port:
38526----\nSIP/2.0
403 Forbidden\r\nVia: SIP/2.0/TCP
10.87.20.5;branch=z9hG4bK-sipXecs-0016a7a00a17b94001516f24875b44f09653;rport=38526\r\nVia:
SIP/2.0/TCP
10.87.20.5;branch=z9hG4bK-sipXecs-0013a7669e28a72d44481cd5a375af980b06~f493f0f98bcc796b31453652fad2d124\r\nVia:
SIP/2.0/UDP
10.87.20.254:5060;branch=z9hG4bK0ee159ab\r\nFrom:
\"Kitchin
Matthew\"
<sip:1...@sipx.voip>;tag=00127f98eaa9004405071fb9-0d560bee\r\nTo:
<sip:916155008...@sipx.voip>\r\nCall-ID:
00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254
<mailto:00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254>
<mailto:00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254
<mailto:00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254>>\r\nCSeq:
102
INVITE\r\nServer: sipXecs/4.0.4 sipXecs/sipxbridge
(Linux)\r\nContact:
<sip:~~id~bri...@10.87.20.5:5090
<http://bri...@10.87.20.5:5090>
<http://bri...@10.87.20.5:5090
<http://10.87.20.5:5090>>>\r\nSupported:
replaces,100rel\r\nReason:
~~id~bridge;cause=213;text=\"Relayed Error
Response\"\r\nContent-Length:
0\r\n\r\n--------------------END--------------------\n"
"2009-11-20T23:11:36.850000Z":20:INCOMING:INFO:nshpbx1.sipx.voip:Thread-16:00000000:sipXbridge:"Read
SIP Message :\n----Remote Host:172.30.209.62---- Port:
5070----\nSIP/2.0
100 Trying\r\nCall-ID:
00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254.0\r\ncseq: 1
INVITE\r\nFrom: \"Kitchin Matthew\"
<sip:1...@pcelbcn0001.dsi.globalipcom.com
<mailto:sip%3a1...@pcelbcn0001.dsi.globalipcom.com>
<mailto:sip%3a1...@pcelbcn0001.dsi.globalipcom.com
<mailto:sip%253a1...@pcelbcn0001.dsi.globalipcom.com>>>;tag=8483786813757111981\r\nTo:
<sip:16155008...@pcelbcn0001.dsi.globalipcom.com
<mailto:sip%3a16155008...@pcelbcn0001.dsi.globalipcom.com>
<mailto:sip%3a16155008...@pcelbcn0001.dsi.globalipcom.com
<mailto:sip%253a16155008...@pcelbcn0001.dsi.globalipcom.com>>;user=phone>\r\nVia:
SIP/2.0/UDP
pcelbcn0001.dsi.globalipcom.com:5080;received=10.87.20.5;branch=z9hG4bKf79a1b7109513b78e1683a1505f003513933\r\nContent-Length:
0\r\n\r\n====================END====================\n"
"2009-11-20T23:11:36.853000Z":21:INCOMING:INFO:nshpbx1.sipx.voip:PipelineThread-0:00000000:sipXbridge:"Read
SIP Message :\n----Remote Host:10.87.20.5---- Port:
38526----\nACK
sip:16155008...@pcelbcn0001.dsi.globalipcom.com:5070
<http://sip:16155008...@pcelbcn0001.dsi.globalipcom.com:5070>
<http://sip:16155008...@pcelbcn0001.dsi.globalipcom.com:5070>
SIP/2.0\r\nRoute:
<sip:10.87.20.5:5090;lr>\r\nContact:
<sip:1...@10.87.20.254:5060;transport=udp;x-sipX-nonat>\r\nFrom:
\"Kitchin Matthew\"
<sip:1...@sipx.voip>;tag=00127f98eaa9004405071fb9-0d560bee\r\nTo:
<sip:916155008...@sipx.voip>\r\nCall-ID:
00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254
<mailto:00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254>
<mailto:00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254
<mailto:00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254>>\r\nCSeq:
102
ACK\r\nMax-Forwards: 20\r\nVia: SIP/2.0/TCP
10.87.20.5;branch=z9hG4bK-sipXecs-0016a7a00a17b94001516f24875b44f09653;rport=38526\r\nContent-Length:
0\r\n\r\n====================END====================\n"
Melting Pot Technologies GMail wrote:
> How are you/they learning the route to your/their
network?
Can you
> ping their .1 address they gave you from the IPPBX?
>
> On Nov 20, 2009, at 6:03 PM,
"mkitchin.pub...@gmail.com <mailto:mkitchin.pub...@gmail.com>
<mailto:mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>>"
> <mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>
<mailto:mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>>> wrote:
>
>> It didn't put anything in a new log file. I've obviously
got some
>> work to do on my end.
>> In a document I gave them several weeks ago, I did
provide
them, the
>> IP of my server.
>>
>> Melting Pot Technologies GMail wrote:
>>> Can you run:
>>>
>>> cd /var/log/sipxpbx
>>>
>>> rm -f ./sipxbridge.log
>>>
>>> Make a test call, and post the results from
sipxbridge.log
>>>
>>> If your a static configuration they are more than
likely
pointing to
>>> a specific address on your end. Did they say anything
about that?
>>>
>>> On Nov 20, 2009, at 5:25 PM,
"mkitchin.pub...@gmail.com <mailto:mkitchin.pub...@gmail.com>
<mailto:mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>>"
>>> <mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>
<mailto:mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>>> wrote:
>>>
>>>> Yes. I just found that under advanced settings. that
seems to have
>>>> gotten rid of that error. Thank you! I still can't
make any
>>>> outbound calls, but hopefully I will be able to
find some
more logs
>>>> showing why. The last entries in my sipxbridge log are
below. My
>>>> inbound calls from Verizon are still set from them to
come in on
>>>> 5060. I have it set to 5080 at the moment so it won;t
conflict with
>>>> my phones attempting to talk to the server. I
assume that
should
>>>> only affect inbound calls, but assuming can make
an ass
out of me.
>>>>
>>>>
"2009-11-20T21:57:48.801000Z":1:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported
>>>> protocol = SSLv2Hello"
>>>>
"2009-11-20T21:57:48.814000Z":2:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported
>>>> protocol = SSLv3"
>>>>
"2009-11-20T21:57:48.814000Z":3:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported
>>>> protocol = TLSv1"
>>>>
"2009-11-20T21:57:49.840000Z":4:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:Gateway:"-------
>>>> REGISTERING--------"
>>>>
>>>>
>>>> Melting Pot Technologies GMail wrote:
>>>>> Did you uncheck register on initialization?
>>>>>
>>>>> On Nov 20, 2009, at 4:24 PM,
"mkitchin.pub...@gmail.com <mailto:mkitchin.pub...@gmail.com>
<mailto:mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>>"
>>>>> <mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>
<mailto:mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>>> wrote:
>>>>>
>>>>>> In case you didn't have enough emails from me,
here is
a little more
>>>>>> info. I put in 123 for the username, so that is
obviously where the
>>>>>> 'sip:1...@pcelbcn0001' entry is coming from. The
service
errors
>>>>>> out if I
>>>>>> don't put in a username and password, but
Verizon isn't
>>>>>> requesting we
>>>>>> use one.
>>>>>>
>>>>>> mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>
<mailto:mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>> wrote:
>>>>>>> Here are some log file entries that appear
relevant to me:
>>>>>>>
>>>>>>>
"2009-11-20T20:08:02.577000Z":5:OUTGOING:INFO:nshpbx1.sipx.voip:main:00000000:sipXbridge:"Sent
>>>>>>>
>>>>>>> SIP Message :\n----Remote
Host:172.30.209.62---- Port:
>>>>>>> 5070----\nREGISTER
sip:pcelbcn0001.munged.munged.com:5070
<http://pcelbcn0001.munged.munged.com:5070>
<http://pcelbcn0001.munged.munged.com:5070>
>>>>>>> SIP/2.0\r\nCall-ID:
>>>>>>> 8d16619b30dd4006b74d20218ff1d...@10.87.20.5
<mailto:8d16619b30dd4006b74d20218ff1d...@10.87.20.5>
<mailto:8d16619b30dd4006b74d20218ff1d...@10.87.20.5
<mailto:8d16619b30dd4006b74d20218ff1d...@10.87.20.5>>\r\nCSeq: 1
>>>>>>> REGISTER\r\nFrom:
>>>>>>> <sip:1...@pcelbcn0001.munged.munged.com
<mailto:sip%3a...@pcelbcn0001.munged.munged.com>
<mailto:sip%3a...@pcelbcn0001.munged.munged.com
<mailto:sip%253a...@pcelbcn0001.munged.munged.com>>>;tag=425578349234274908\r\nTo:
>>>>>>>
>>>>>>> <sip:1...@pcelbcn0001.munged.munged.com
<mailto:sip%3a...@pcelbcn0001.munged.munged.com>
<mailto:sip%3a...@pcelbcn0001.munged.munged.com
<mailto:sip%253a...@pcelbcn0001.munged.munged.com>>>\r\nVia:
SIP/2.0/UDP
>>>>>>>
pcelbcn0001.munged.munged.com:5060;branch=z9hG4bK4bc3200f1f2720efc3aa25861a8eaa62333134\r\nMax-Forwards:
>>>>>>>
>>>>>>> 70\r\nUser-Agent: sipXecs/4.0.4 sipXecs/sipxbridge
>>>>>>> (Linux)\r\nAllow:
>>>>>>> INVITE,BYE,ACK,CANCEL,OPTIONS\r\nRoute:
>>>>>>>
<sip:172.30.209.62:5070;transport=udp;lr>\r\nContact:
>>>>>>>
<sip:1...@pcelbcn0001.munged.munged.com:5060;transport=udp>\r\nExpires:
>>>>>>>
>>>>>>> 600\r\nContent-Length:
>>>>>>>
0\r\n\r\n--------------------END--------------------\n"
>>>>>>>
"2009-11-20T20:08:02.613000Z":6:INCOMING:INFO:nshpbx1.sipx.voip:Thread-13:00000000:sipXbridge:"Read
>>>>>>>
>>>>>>> SIP Message :\n----Remote
Host:172.30.209.62---- Port:
>>>>>>> 5070----\nSIP/2.0 403 Forbidden\r\nVia: SIP/2.0/UDP
>>>>>>>
pcelbcn0001.munged.munged.com:5060;received=10.87.20.5;branch=z9hG4bK4bc3200f1f2720efc3aa25861a8eaa62333134\r\nCall-ID:
>>>>>>>
>>>>>>> 8d16619b30dd4006b74d20218ff1d...@10.87.20.5
<mailto:8d16619b30dd4006b74d20218ff1d...@10.87.20.5>
<mailto:8d16619b30dd4006b74d20218ff1d...@10.87.20.5
<mailto:8d16619b30dd4006b74d20218ff1d...@10.87.20.5>>\r\nCSeq: 1
>>>>>>> REGISTER\r\nFrom:
>>>>>>> <sip:1...@pcelbcn0001.munged.munged.com
<mailto:sip%3a...@pcelbcn0001.munged.munged.com>
<mailto:sip%3a...@pcelbcn0001.munged.munged.com
<mailto:sip%253a...@pcelbcn0001.munged.munged.com>>>;tag=425578349234274908\r\nTo:
>>>>>>>
>>>>>>> <sip:1...@pcelbcn0001.munged.munged.com
<mailto:sip%3a...@pcelbcn0001.munged.munged.com>
<mailto:sip%3a...@pcelbcn0001.munged.munged.com
<mailto:sip%253a...@pcelbcn0001.munged.munged.com>>>;tag=aprqngfrt-gjiai91000020\r\nContent-Length:
>>>>>>>
>>>>>>>
0\r\n\r\n====================END====================\n"
>>>>>>> <?xml version="1.0" encoding="UTF-8"
standalone="no"?>
>>>>>>> <!DOCTYPE log SYSTEM "logger.dtd">
>>>>>>> <log>
>>>>>>> </log>
>>>>>>>
"2009-11-20T20:08:12.601000Z":1:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported
>>>>>>>
>>>>>>> protocol = SSLv2Hello"
>>>>>>>
"2009-11-20T20:08:12.602000Z":2:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported
>>>>>>>
>>>>>>> protocol = SSLv3"
>>>>>>>
"2009-11-20T20:08:12.602000Z":3:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported
>>>>>>>
>>>>>>> protocol = TLSv1"
>>>>>>>
"2009-11-20T20:08:12.683000Z":4:JAVA:ERR:nshpbx1.sipx.voip:main:00000000:sipXbridge:"gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1080)
>>>>>>>
>>>>>>> [Invalid argument address = 10.87.20.5 port = 5060
transport =
>>>>>>> udp]"
>>>>>>>
"2009-11-20T20:08:12.686000Z":5:JAVA:ERR:nshpbx1.sipx.voip:main:00000000:Gateway:"Cannot
>>>>>>>
>>>>>>> initialize gateway"
>>>>>>> javax.sip.InvalidArgumentException: Address
already in use
>>>>>>>
>>>>>>>
>>>>>>> mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>
<mailto:mkitchin.pub...@gmail.com
<mailto:mkitchin.pub...@gmail.com>> wrote:
>>>>>>>> This was an inevitable question from me. I
need some help
>>>>>>>> connecting
>>>>>>>> to Verizon SIP over a private DS3. There is no
firewall or NAT
>>>>>>>> involved. The information they gave me is below.
>>>>>>>>
>>>>>>>> From Verizon:
>>>>>>>> Inbound calls will route from the
172.30.9.0/24 <http://172.30.9.0/24>
<http://172.30.9.0/24> port 5060
>>>>>>>> network and
>>>>>>>> you should be able to ping 172.30.9.1. This
is the only
>>>>>>>> address you
>>>>>>>> will be able to ping for security reasons.
>>>>>>>> For outbound calls please configure the SIP target
(to the VzB
>>>>>>>> network) to one of the settings below.
>>>>>>>> IP: 172.30.209.62 port: 5070
>>>>>>>> OR
>>>>>>>> FQDN: pcelbcn0001.munged.munged.com
<http://pcelbcn0001.munged.munged.com>
<http://pcelbcn0001.munged.munged.com>
>>>>>>>>
>>>>>>>> I'm using the sipexec server as the SBC. It is at
10.87.20.5. I
>>>>>>>> have
>>>>>>>> tried to translate this into all the correct
fields
on the
>>>>>>>> configuration guide here:
>>>>>>>>
http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration
>>>>>>>>
>>>>>>>>
>>>>>>>> No matter what I try, The Sip Trunking service
fails
to start with
>>>>>>>> the 'Address already in use' error below. I
googled
several of the
>>>>>>>> lines, and I found some bug reports and other
writeups that didn't
>>>>>>>> appear to relate to my problem. I cleared one
other
error by
>>>>>>>> putting
>>>>>>>> in a fake username and password under ITSP
account. I
don't
>>>>>>>> have an
>>>>>>>> username and password. I would assume that is
because
this is a
>>>>>>>> private connection. As you can see, I have
received
minimal
>>>>>>>> information from Verizon. I also have no NAT
or firewall
>>>>>>>> involved, so
>>>>>>>> several of the configuration screens regarding NAT
don't really
>>>>>>>> pertain to me, but I had to put in a value of some
sort. On
>>>>>>>> System,
>>>>>>>> Servers, NAT, Public IP address for example, I
had to put
>>>>>>>> something,
>>>>>>>> so I put 10.87.20.5. Verizon has performed a
miracle
in their
>>>>>>>> minds
>>>>>>>> by simply agreeing to work with a 'vendorless open
source PBX',
>>>>>>>> and
>>>>>>>> we are supposed to have their Interop test
with wireshark
>>>>>>>> captures on
>>>>>>>> Monday. I need to do anything possible to get this
working by
>>>>>>>> then.
>>>>>>>> With the information I have, can someone help me
figure out
>>>>>>>> exactly
>>>>>>>> what values should be put where in the various
config
screens?
>>>>>>>> A few
>>>>>>>> are obvious, but a few aren't for me at least
given
give that
>>>>>>>> there
>>>>>>>> is no firewall, NAT or ITSP account.
>>>>>>>>
>>>>>>>> Thanks a ton,
>>>>>>>> Matthew
>>>>>>>>
>>>>>>>> javax.sip.InvalidArgumentException: Address
already
in use
>>>>>>>> at
>>>>>>>>
gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1083)
>>>>>>>>
>>>>>>>>
>>>>>>>> at
>>>>>>>>
org.sipfoundry.sipxbridge.Gateway.initializeSipListeningPoints(Gateway.java:540)
>>>>>>>>
>>>>>>>>
>>>>>>>> at
org.sipfoundry.sipxbridge.Gateway.start(Gateway.java:1000)
>>>>>>>> at
org.sipfoundry.sipxbridge.Gateway.main(Gateway.java:1353)
>>>>>>>> Caused by: java.io.IOException: Address
already in use
>>>>>>>> at
>>>>>>>>
gov.nist.javax.sip.stack.UDPMessageProcessor.<init>(UDPMessageProcessor.java:130)
>>>>>>>>
>>>>>>>>
>>>>>>>> at
>>>>>>>>
gov.nist.javax.sip.stack.SIPTransactionStack.createMessageProcessor(SIPTransactionStack.java:1890)
>>>>>>>>
>>>>>>>>
>>>>>>>> at
>>>>>>>>
gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1064)
>>>>>>>>
>>>>>>>>
>>>>>>>> ... 3 more
>>>>>>>> SipXbridge : Exception caught while running
>>>>>>>> org.sipfoundry.sipxbridge.SipXbridgeException:
Cannot
>>>>>>>> initialize gateway
>>>>>>>> at
>>>>>>>>
org.sipfoundry.sipxbridge.Gateway.initializeSipListeningPoints(Gateway.java:598)
>>>>>>>>
>>>>>>>>
>>>>>>>> at
org.sipfoundry.sipxbridge.Gateway.start(Gateway.java:1000)
>>>>>>>> at
org.sipfoundry.sipxbridge.Gateway.main(Gateway.java:1353)
>>>>>>>> Caused by: javax.sip.InvalidArgumentException:
Address already
>>>>>>>> in use
>>>>>>>> at
>>>>>>>>
gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1083)
>>>>>>>>
>>>>>>>>
>>>>>>>> at
>>>>>>>>
org.sipfoundry.sipxbridge.Gateway.initializeSipListeningPoints(Gateway.java:540)
>>>>>>>>
>>>>>>>>
>>>>>>>> ... 2 more
>>>>>>>> Caused by: java.io.IOException: Address
already in use
>>>>>>>> at
>>>>>>>>
gov.nist.javax.sip.stack.UDPMessageProcessor.<init>(UDPMessageProcessor.java:130)
>>>>>>>>
>>>>>>>>
>>>>>>>> at
>>>>>>>>
gov.nist.javax.sip.stack.SIPTransactionStack.createMessageProcessor(SIPTransactionStack.java:1890)
>>>>>>>>
>>>>>>>>
>>>>>>>> at
>>>>>>>>
gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1064)
>>>>>>>>
>>>>>>>>
>>>>>>>> ... 3 more
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
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>>>>
>>
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