407 response to inbound ITSP calls

[Kurt Albershardt]

I'm seeing 407 (Proxy Authentication Required) returned to incoming invites from Vitelity. We have a no-registration (IP authentication) relationship with them so it's possible I've missed something in the setup that would allow unauthenticated invites from their IP.


This is the response sent to sipxbridge:

Time: 2012-07-20T16:48:16.097000Z
Frame: 4 sipxbridge.xml:354
Source: 192.168.X.24:5060
Dest: sipx.domain.com-sipXbridge

SIP/2.0 407 Proxy Authentication Required
Record-Route: <sip:192.168.X.24:5060;lr;sipXecs-CallDest=LD;x-sipX-done>
From: "[phone]" <sip:***@66.241.X.X>;tag=2014970589
To: <sip:***@sipx.domain.com>;tag=1R5cwk
Call-ID: ***@66.241.X.X-0
CSeq: 102 INVITE
Via: SIP/2.0/TCP 192.168.X.24:5090;branch=z9hG4bK11c84e5a79571525f8a761d6bcaf2a4c393639;sipxecs-id=3164ecee
Proxy-Authenticate: Digest realm="sipx.domain.com",nonce="d98c3b153f9c64d522cd72f9a7c28fde50098bd0",qop="auth"
Server: sipXecs/4.4.0 sipXecs/sipXproxy (Linux)
Date: Fri, 20 Jul 2012 16:48:16 GMT
Content-Length: 0

...which sends this to Vitelity:


Time: 2012-07-20T16:48:16.103000Z
Frame: 6 sipxbridge.xml:365
Source: sipx.domain.com-sipXbridge
Dest: 66.241.X.X:5060

SIP/2.0 407 Proxy Authentication required
Via: SIP/2.0/UDP 66.241.X.X:5060;branch=z9hG4bK17b48c37;rport=5060;received=66.241.X.X
From: "[phone]" <sip:***@66.241.X.X>;tag=as4b214e18
To: <sip:***@70.57.247.39:5060>
Call-ID: ***@66.241.X.X
CSeq: 102 INVITE
Server: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
Supported: replaces
Record-Route: <sip:192.168.X.24:5060;lr;sipXecs-CallDest=LD;x-sipX-done>
Proxy-Authenticate: Digest realm="sipx.domain.com",nonce="d98c3b153f9c64d522cd72f9a7c28fde50098bd0",qop="auth"
Date: Fri, 20 Jul 2012 16:48:16 GMT
Contact: <sip:70.57.247.39:5080>
Content-Length: 0

[Tony Graziano]

Then you would have set this up using a template in sipx as a "trunk" and
choosing "bandwidth.com" as the template and changing the addresses to
match vitelity. I do not think it will work unless vitelity is sending the
inbound invite onport 5080, which I cannot see they are due to the limited
information you have provided.

If the invites are coming from vitelity on port 5060, it WOULD act as an
unauthorized call.

Change those two things. Use the bandwidth.com template and have vitelity
send to port 5080.

Post by Kurt Albershardt
I'm seeing 407 (Proxy Authentication Required) returned to incoming
invites from Vitelity. We have a no-registration (IP authentication)
relationship with them so it's possible I've missed something in the setup
that would allow unauthenticated invites from their IP.
Time: 2012-07-20T16:48:16.097000Z
Frame: 4 sipxbridge.xml:354
Source: 192.168.X.24:5060
Dest: sipx.domain.com-sipXbridge
SIP/2.0 407 Proxy Authentication Required
Record-Route: <sip:192.168.X.24:5060;lr;sipXecs-CallDest=LD;x-sipX-done>
CSeq: 102 INVITE
Via: SIP/2.0/TCP
192.168.X.24:5090;branch=z9hG4bK11c84e5a79571525f8a761d6bcaf2a4c393639;sipxecs-id=3164ecee
Proxy-Authenticate: Digest realm="sipx.domain.com
",nonce="d98c3b153f9c64d522cd72f9a7c28fde50098bd0",qop="auth"
Server: sipXecs/4.4.0 sipXecs/sipXproxy (Linux)
Date: Fri, 20 Jul 2012 16:48:16 GMT
Content-Length: 0
Time: 2012-07-20T16:48:16.103000Z
Frame: 6 sipxbridge.xml:365
Source: sipx.domain.com-sipXbridge
Dest: 66.241.X.X:5060
SIP/2.0 407 Proxy Authentication required
Via: SIP/2.0/UDP
66.241.X.X:5060;branch=z9hG4bK17b48c37;rport=5060;received=66.241.X.X
CSeq: 102 INVITE
Server: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
Supported: replaces
Record-Route: <sip:192.168.X.24:5060;lr;sipXecs-CallDest=LD;x-sipX-done>
Proxy-Authenticate: Digest realm="sipx.domain.com
",nonce="d98c3b153f9c64d522cd72f9a7c28fde50098bd0",qop="auth"
Date: Fri, 20 Jul 2012 16:48:16 GMT
Contact: <sip:70.57.247.39:5080>
Content-Length: 0
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[Kurt Albershardt]

Vitelity is sending to 5060 -- when I asked about 5080 they said:

"We could change it on our end but do not recommend this. In addition if we do you will be limited to 30 channels and this amount will not be able to be increased in the future."

30 channels should be more than adequate for the next year or so at this site anyway.

I'm curious about the suggestion to use a sipxbridge instance (rather than an unmanaged gateway as was suggested yesterday) and choosing bandwidth.com instead of Vitelity as the template?

thanks...

Then you would have set this up using a template in sipx as a "trunk" and choosing "bandwidth.com" as the template and changing the addresses to match vitelity. I do not think it will work unless vitelity is sending the inbound invite onport 5080, which I cannot see they are due to the limited information you have provided.
If the invites are coming from vitelity on port 5060, it WOULD act as an unauthorized call.
Change those two things. Use the bandwidth.com template and have vitelity send to port 5080.
I'm seeing 407 (Proxy Authentication Required) returned to incoming invites from Vitelity. We have a no-registration (IP authentication) relationship with them so it's possible I've missed something in the setup that would allow unauthenticated invites from their IP.
Time: 2012-07-20T16:48:16.097000Z
Frame: 4 sipxbridge.xml:354
Source: 192.168.X.24:5060
Dest: sipx.domain.com-sipXbridge
SIP/2.0 407 Proxy Authentication Required
Record-Route: <sip:192.168.X.24:5060;lr;sipXecs-CallDest=LD;x-sipX-done>
CSeq: 102 INVITE
Via: SIP/2.0/TCP 192.168.X.24:5090;branch=z9hG4bK11c84e5a79571525f8a761d6bcaf2a4c393639;sipxecs-id=3164ecee
Proxy-Authenticate: Digest realm="sipx.domain.com",nonce="d98c3b153f9c64d522cd72f9a7c28fde50098bd0",qop="auth"
Server: sipXecs/4.4.0 sipXecs/sipXproxy (Linux)
Date: Fri, 20 Jul 2012 16:48:16 GMT
Content-Length: 0
Time: 2012-07-20T16:48:16.103000Z
Frame: 6 sipxbridge.xml:365
Source: sipx.domain.com-sipXbridge
Dest: 66.241.X.X:5060
SIP/2.0 407 Proxy Authentication required
Via: SIP/2.0/UDP 66.241.X.X:5060;branch=z9hG4bK17b48c37;rport=5060;received=66.241.X.X
CSeq: 102 INVITE
Server: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
Supported: replaces
Record-Route: <sip:192.168.X.24:5060;lr;sipXecs-CallDest=LD;x-sipX-done>
Proxy-Authenticate: Digest realm="sipx.domain.com",nonce="d98c3b153f9c64d522cd72f9a7c28fde50098bd0",qop="auth"
Date: Fri, 20 Jul 2012 16:48:16 GMT
Contact: <sip:70.57.247.39:5080>
Content-Length: 0
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[Tony Graziano]

I think if you listen to meL

have them send on port 5080
use sipxbridge to anchor the calls (otherwise you will not be able to
transfer it)

it will work with vitelty, regardless of their inferences.

Post by Kurt Albershardt
"We could change it on our end but do not recommend this. In addition
if we do you will be limited to 30 channels and this amount will not
be able to be increased in the future."
30 channels should be more than adequate for the next year or so at this site anyway.
I'm curious about the suggestion to use a sipxbridge instance (rather than
an unmanaged gateway as was suggested yesterday) and choosing
bandwidth.com instead of Vitelity as the template?
thanks...
Then you would have set this up using a template in sipx as a "trunk" and
choosing "bandwidth.com" as the template and changing the addresses to
match vitelity. I do not think it will work unless vitelity is sending the
inbound invite onport 5080, which I cannot see they are due to the limited
information you have provided.
If the invites are coming from vitelity on port 5060, it WOULD act as an unauthorized call.
Change those two things. Use the bandwidth.com template and have vitelity
send to port 5080.

Post by Kurt Albershardt
I'm seeing 407 (Proxy Authentication Required) returned to incoming
invites from Vitelity. We have a no-registration (IP authentication)
relationship with them so it's possible I've missed something in the setup
that would allow unauthenticated invites from their IP.
Time: 2012-07-20T16:48:16.097000Z
Frame: 4 sipxbridge.xml:354
Source: 192.168.X.24:5060
Dest: sipx.domain.com-sipXbridge
SIP/2.0 407 Proxy Authentication Required
Record-Route: <sip:192.168.X.24:5060;lr;sipXecs-CallDest=LD;x-sipX-done>
CSeq: 102 INVITE
Via: SIP/2.0/TCP
192.168.X.24:5090;branch=z9hG4bK11c84e5a79571525f8a761d6bcaf2a4c393639;sipxecs-id=3164ecee
Proxy-Authenticate: Digest realm="sipx.domain.com
",nonce="d98c3b153f9c64d522cd72f9a7c28fde50098bd0",qop="auth"
Server: sipXecs/4.4.0 sipXecs/sipXproxy (Linux)
Date: Fri, 20 Jul 2012 16:48:16 GMT
Content-Length: 0
Time: 2012-07-20T16:48:16.103000Z
Frame: 6 sipxbridge.xml:365
Source: sipx.domain.com-sipXbridge
Dest: 66.241.X.X:5060
SIP/2.0 407 Proxy Authentication required
Via: SIP/2.0/UDP
66.241.X.X:5060;branch=z9hG4bK17b48c37;rport=5060;received=66.241.X.X
CSeq: 102 INVITE
Server: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
Supported: replaces
Record-Route: <sip:192.168.X.24:5060;lr;sipXecs-CallDest=LD;x-sipX-done>
Proxy-Authenticate: Digest realm="sipx.domain.com
",nonce="d98c3b153f9c64d522cd72f9a7c28fde50098bd0",qop="auth"
Date: Fri, 20 Jul 2012 16:48:16 GMT
Contact: <sip:70.57.247.39:5080>
Content-Length: 0
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~~~~~~~~~~~~~~~~~~
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Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~
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[Kurt Albershardt]

Post by Tony Graziano
I think if you listen to meL

Assume you meant 'me' there, and I am!

Post by Tony Graziano
use sipxbridge to anchor the calls (otherwise you will not be able to transfer it)

Aha, that makes sense.


I was just curious about what in the bandwidth.com template made it more appropriate to use than the Vitelity template.

Should I also choose the bandwidth.com template for outbound calls to Vitelity with this non-registering config?


thanks!

[Tony Graziano]

I refer to the bandwidth.com template because it has all the correct
settings in there for non-registration (ip based only) and works fine in
such cases.

Post by Kurt Albershardt

Post by Tony Graziano
I think if you listen to meL

Assume you meant 'me' there, and I am!

Post by Tony Graziano
use sipxbridge to anchor the calls (otherwise you will not be able to

transfer it)
Aha, that makes sense.
I was just curious about what in the bandwidth.com template made it more
appropriate to use than the Vitelity template.
Should I also choose the bandwidth.com template for outbound calls to
Vitelity with this non-registering config?
thanks!
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List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
sip: ***@voice.myitdepartment.net
Fax: [phone]
~~~~~~~~~~~~~~~~~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!
<http://sipxcolab2013.eventbrite.com/?discount=tony2013>

[Kurt Albershardt]

Gotcha - thanks.

I did notice a whole lot of extraneous consumer-like config junk in the Vitelity template.

I refer to the bandwidth.com template because it has all the correct settings in there for non-registration (ip based only) and works fine in such cases.

Post by Tony Graziano
I think if you listen to meL

Assume you meant 'me' there, and I am!

Post by Tony Graziano
use sipxbridge to anchor the calls (otherwise you will not be able to transfer it)

Aha, that makes sense.
I was just curious about what in the bandwidth.com template made it more appropriate to use than the Vitelity template.
Should I also choose the bandwidth.com template for outbound calls to Vitelity with this non-registering config?
thanks!
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--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
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Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013!
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[Michael Picher]

Version 4.6 should have port 5060 consolidation for sip trunking... I'm
still under the assumption this made it...

Post by Kurt Albershardt
Gotcha - thanks.
I did notice a whole lot of extraneous consumer-like config junk in the Vitelity template.
I refer to the bandwidth.com template because it has all the correct
settings in there for non-registration (ip based only) and works fine in
such cases.

Post by Kurt Albershardt

Post by Tony Graziano
I think if you listen to meL

Assume you meant 'me' there, and I am!

Post by Tony Graziano
use sipxbridge to anchor the calls (otherwise you will not be able to

transfer it)
Aha, that makes sense.
I was just curious about what in the bandwidth.com template made it more
appropriate to use than the Vitelity template.
Should I also choose the bandwidth.com template for outbound calls to
Vitelity with this non-registering config?
thanks!
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List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
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Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013!
<http://sipxcolab2013.eventbrite.com/?discount=tony2013>
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------------------------------------------------------------------------------------------------------------
There are 10 kinds of people in the world, those who understand binary and
those who don't.

[Tony Graziano]

And here I am thinking it didn't make it in.

Post by Michael Picher
Version 4.6 should have port 5060 consolidation for sip trunking... I'm
still under the assumption this made it...

Post by Kurt Albershardt
Gotcha - thanks.
I did notice a whole lot of extraneous consumer-like config junk in the
Vitelity template.
I refer to the bandwidth.com template because it has all the correct
settings in there for non-registration (ip based only) and works fine in
such cases.

Post by Kurt Albershardt

Post by Tony Graziano
I think if you listen to meL

Assume you meant 'me' there, and I am!

Post by Tony Graziano
use sipxbridge to anchor the calls (otherwise you will not be able to

transfer it)
Aha, that makes sense.
I was just curious about what in the bandwidth.com template made it
more appropriate to use than the Vitelity template.
Should I also choose the bandwidth.com template for outbound calls to
Vitelity with this non-registering config?
thanks!
_______________________________________________
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List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: [phone]
Fax: [phone]
~~~~~~~~~~~~~~~~~~
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~~~~~~~~~~~~~~~~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013!
<http://sipxcolab2013.eventbrite.com/?discount=tony2013>
Telephone: [phone]
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300 Brickstone Square****
Suite 201****
Andover, MA. 01810
O.978-296-1005 X2015
M.207-956-0262
@mpicher <http://twitter.com/mpicher>
linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro>
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------------------------------------------------------------------------------------------------------------
There are 10 kinds of people in the world, those who understand binary and
those who don't.
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[Kurt Albershardt]

Ack - sorry about the large attachment.

Getting closer - have AON configured (thanks, Tony) on pfSense and am receiving signaling on 5080. Now sipx is responding to invites with 302 first and then 407. There's obviously a lot of data here -- I can forward details of any message(s) or the entire XML file.
<PastedGraphic-2.tiff>
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[Tony Graziano]

A graphic won't help anyone troubleshoot it. Post the trace file.

Post by Kurt Albershardt
Ack - sorry about the large attachment.

Getting closer - have AON configured (thanks, Tony) on pfSense and am

receiving signaling on 5080. Now sipx is responding to invites with 302
first and then 407. There's obviously a lot of data here -- I can forward
details of any message(s) or the entire XML file.

<PastedGraphic-2.tiff>
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[Kurt Albershardt]

Attached - thanks.

Post by Tony Graziano
A graphic won't help anyone troubleshoot it. Post the trace file.

[Kurt Albershardt]

Anyone have insights from this trace?

Thanks~

Post by Kurt Albershardt
Attached - thanks.
<merged.xml>

Post by Tony Graziano
A graphic won't help anyone troubleshoot it. Post the trace file.

[Tony Graziano]

Explain what user or service has a did or alias of 5755195606. That is what
is. Being sent in the invite. Normally you would dial this 10 digit number
internally to make sure the input is correct before trying against an itsp.

Please explain the call flow.

Post by Kurt Albershardt
Anyone have insights from this trace?
Thanks~
Attached - thanks.
<merged.xml>
A graphic won't help anyone troubleshoot it. Post the trace file.
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[Kurt Albershardt]

Thanks for the wake-up!
Found a typo in the alias I setup for testing. Now inbound calls are working.

Back to 4.6 for another round of testing...

Explain what user or service has a did or alias of 5755195606. That is what is. Being sent in the invite. Normally you would dial this 10 digit number internally to make sure the input is correct before trying against an itsp.

[Tony Graziano]

I have been only on a mobile device for the last week. My mobile device was
able to view the XML file and told me the call came in but was "trying" and
failing and probably why.

Post by Kurt Albershardt
Thanks for the wake-up!
Found a typo in the alias I setup for testing. Now inbound calls are working.
Back to 4.6 for another round of testing...
Explain what user or service has a did or alias of 5755195606. That is
what is. Being sent in the invite. Normally you would dial this 10 digit
number internally to make sure the input is correct before trying against
an itsp.
_______________________________________________
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[Kurt Albershardt]

Getting closer - have AON configured (thanks, Tony) on pfSense and am receiving signaling on 5080. Now sipx is responding to invites with 302 first and then 407. There's obviously a lot of data here -- I can forward details of any message(s) or the entire XML file.