Re: [sipx-users] we had major mailing list outage today
testing attachments On Tue, Sep 4, 2012 at 10:05 PM, Todd Hodgen thod...@frontier.com wrote: Are attachments going through? I see my response to your email went through. However, one that I had sent that has a small attachment hasn't yet. It was sent prior to my response to your test message. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler Sent: Tuesday, September 04, 2012 6:44 PM To: sipx-users Subject: [sipx-users] we had major mailing list outage today this is a test to see if ML is back. Issue: my fault, dumb config mistake. ___ 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/ configure.ac Description: Binary data ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] we had major mailing list outage today
Getting attachs On Wed, Sep 5, 2012 at 11:04 AM, Douglas Hubler dhub...@ezuce.com wrote: testing attachments On Tue, Sep 4, 2012 at 10:05 PM, Todd Hodgen thod...@frontier.com wrote: Are attachments going through? I see my response to your email went through. However, one that I had sent that has a small attachment hasn't yet. It was sent prior to my response to your test message. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler Sent: Tuesday, September 04, 2012 6:44 PM To: sipx-users Subject: [sipx-users] we had major mailing list outage today this is a test to see if ML is back. Issue: my fault, dumb config mistake. ___ 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 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/
Re: [sipx-users] we had major mailing list outage today
same here, got it On Wed, Sep 5, 2012 at 11:08 AM, George Niculae geo...@ezuce.com wrote: Getting attachs On Wed, Sep 5, 2012 at 11:04 AM, Douglas Hubler dhub...@ezuce.com wrote: testing attachments On Tue, Sep 4, 2012 at 10:05 PM, Todd Hodgen thod...@frontier.com wrote: Are attachments going through? I see my response to your email went through. However, one that I had sent that has a small attachment hasn't yet. It was sent prior to my response to your test message. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler Sent: Tuesday, September 04, 2012 6:44 PM To: sipx-users Subject: [sipx-users] we had major mailing list outage today this is a test to see if ML is back. Issue: my fault, dumb config mistake. ___ 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 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 list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS handshake fails in 4.6
On Tue, Sep 4, 2012 at 3:26 PM, John Lightfoot j...@vizhn.com wrote: I thought I'd put some effort into getting 4.6 to work with Exchange Online UM which requires TLS transport with a non-self-signed SSL certificate. Right out of the gate, the TLS handshake fails when Exchange requests Sipx's client certificate and Sipx returns a handshake message with the cert empty. There's nothing logged that indicates Sipx doesn't like the request, signing authority (MS's intermediate authority certs are loaded in Sipx) or the like. The sipxbridge.log only indicates that the far end rudely tears the session down but nothing else. I had it configured with 4.4 with the same cert and same Exchange config and the cert exchange happened as expected. I'm a little unclear how to turn the TLS logging to debug or what John, no takers as far as developers responding, but I'm very thrilled you're testing. sipXbridge's TLS implementation hasn't changed AFAIK, but cert. management certainly has. Can you verify proper cert is delivered? ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS handshake fails in 4.6
On Wed, Sep 5, 2012 at 5:55 PM, Douglas Hubler dhub...@ezuce.com wrote: On Tue, Sep 4, 2012 at 3:26 PM, John Lightfoot j...@vizhn.com wrote: I thought I'd put some effort into getting 4.6 to work with Exchange Online UM which requires TLS transport with a non-self-signed SSL certificate. Right out of the gate, the TLS handshake fails when Exchange requests Sipx's client certificate and Sipx returns a handshake message with the cert empty. There's nothing logged that indicates Sipx doesn't like the request, signing authority (MS's intermediate authority certs are loaded in Sipx) or the like. The sipxbridge.log only indicates that the far end rudely tears the session down but nothing else. I had it configured with 4.4 with the same cert and same Exchange config and the cert exchange happened as expected. I'm a little unclear how to turn the TLS logging to debug or what John, no takers as far as developers responding, but I'm very thrilled you're testing. sipXbridge's TLS implementation hasn't changed AFAIK, but cert. management certainly has. Can you verify proper cert is delivered? Also please provide a sipxbridge logs on debug (snapshot would be better as we could see the configuration). Habe you set Transport Protocol on TLS or Auto? (it could be setting not replicating issue) George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS handshake fails in 4.6
In response to Doug's question, I've tested with openssl from the command line using the same certs and the TLS negotiates fine. I also had it configured pretty much identically in 4.4 and the Sipx cert was sent. Oh and I tried with the self-signed cert as well and got the same results. As for transport, I have the gateway configured with TLS as the transport. Since TLS tries to negotiate, it would appear I at least have that part correct. I'm not sure how to set logging on Sipxbridge to debug. If I can get some guidance on that, I'll upload ASAP. John ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS handshake fails in 4.6
On Wed, Sep 5, 2012 at 2:54 PM, John Lightfoot j...@vizhn.com wrote: In response to Doug's question, I've tested with openssl from the command line using the same certs and the TLS negotiates fine. I also had it configured pretty much identically in 4.4 and the Sipx cert was sent. Oh and I tried with the self-signed cert as well and got the same results. As for transport, I have the gateway configured with TLS as the transport. Since TLS tries to negotiate, it would appear I at least have that part correct. I'm not sure how to set logging on Sipxbridge to debug. If I can get some guidance on that, I'll upload ASAP. John Sure, go to Devices SIP Trunk SBCs, click on name (should be sipXbridge-1) then Logging leve section. I think I know why problem lies and this is related to setting replication, could you look into /etc/sipxpbx/sipxbridge.xml file and check if itsp-transport for that account reflects TLS selection? George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS handshake fails in 4.6
systemlogging levelsadvancedtrunking (which is sipxbridge) set to debug and restart services as prompted On Wed, Sep 5, 2012 at 7:54 AM, John Lightfoot j...@vizhn.com wrote: In response to Doug's question, I've tested with openssl from the command line using the same certs and the TLS negotiates fine. I also had it configured pretty much identically in 4.4 and the Sipx cert was sent. Oh and I tried with the self-signed cert as well and got the same results. As for transport, I have the gateway configured with TLS as the transport. Since TLS tries to negotiate, it would appear I at least have that part correct. I'm not sure how to set logging on Sipxbridge to debug. If I can get some guidance on that, I'll upload ASAP. John ___ 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: 434.465.6833 ~~ 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 -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS handshake fails in 4.6
oops. I sent that for 4.4, for 4.6 refer to what george sent On Wed, Sep 5, 2012 at 11:57 AM, George Niculae geo...@ezuce.com wrote: On Wed, Sep 5, 2012 at 2:54 PM, John Lightfoot j...@vizhn.com wrote: In response to Doug's question, I've tested with openssl from the command line using the same certs and the TLS negotiates fine. I also had it configured pretty much identically in 4.4 and the Sipx cert was sent. Oh and I tried with the self-signed cert as well and got the same results. As for transport, I have the gateway configured with TLS as the transport. Since TLS tries to negotiate, it would appear I at least have that part correct. I'm not sure how to set logging on Sipxbridge to debug. If I can get some guidance on that, I'll upload ASAP. John Sure, go to Devices SIP Trunk SBCs, click on name (should be sipXbridge-1) then Logging leve section. I think I know why problem lies and this is related to setting replication, could you look into /etc/sipxpbx/sipxbridge.xml file and check if itsp-transport for that account reflects TLS selection? George ___ 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: 434.465.6833 ~~ 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 -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[sipx-users] 4.6 Polycom registration problem
I'm thinking I am missing something stupid here but didn't figure it out quite yet. I have been unsuccessful at getting a Polycom 335 to register with a 4.6 test server. Attached is an interesting snippet from sipregistrar.log that sounds like some kind of database lookup problem perhaps. This is the phone attempting to register with line 1 extension 1910. The user was created and assigned to the phone with virtually no modification from the defaults. The phone device was created by the auto-provision service. In fact the phone registers just fine before having a line assigned to it. Here is what was done to get to this point: * Install new 4.6 server from the ISO followed by a yum update * Created a new test domain and made this server the 1st server. DNS entries were created for the test domain and the DNS Advisor reports all is well. * Added all of the Telephony Services, Instant Messaging, and Phone Auto-Provisioning services * Boot up a Polycom 335 with a cleared config in the test domain. It found the server, was auto-provisioned, and was registered. * Changed voicemail dial plan to 4 digits and extension 1900 * Created a user 1910 with password and PIN * Assigned user 1910 to line 1 on the polycom device and did a Send Profile * Phone reboots, comes up with 1910 on the display for Line 1 but is not able to register * Changed LOG level for SipRegistrar to INFO level and obtain the attached snippet during a phone reboot. * Blew away the server and tried a 2nd round just in case Same results. Thoughts? Tony Wyland Messiah College wyl...@messiah.edumailto:wyl...@messiah.edu sipregistrar-1910.log Description: sipregistrar-1910.log ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.6 Polycom registration problem
Are you running above 3.2.7 firmware on the phone? -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ 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! On Sep 5, 2012 1:15 PM, Wyland, Tony wyl...@messiah.edu wrote: I’m thinking I am missing something stupid here but didn’t figure it out quite yet. I have been unsuccessful at getting a Polycom 335 to register with a 4.6 test server. Attached is an interesting snippet from sipregistrar.log that sounds like some kind of database lookup problem perhaps. This is the phone attempting to register with line 1 extension 1910. The user was created and assigned to the phone with virtually no modification from the defaults. The phone device was created by the auto-provision service. In fact the phone registers just fine before having a line assigned to it. Here is what was done to get to this point: ** ** **· **Install new 4.6 server from the ISO followed by a yum update **· **Created a new test domain and made this server the 1stserver. DNS entries were created for the test domain and the DNS Advisor reports all is well. **· **Added all of the Telephony Services, Instant Messaging, and Phone Auto-Provisioning services **· **Boot up a Polycom 335 with a cleared config in the test domain. It found the server, was auto-provisioned, and was registered.*** * **· **Changed voicemail dial plan to 4 digits and extension 1900** ** **· **Created a user 1910 with password and PIN **· **Assigned user 1910 to line 1 on the polycom device and did a Send Profile **· **Phone reboots, comes up with 1910 on the display for Line 1 but is not able to register **· **Changed LOG level for SipRegistrar to INFO level and obtain the attached snippet during a phone reboot. **· **Blew away the server and tried a 2nd round just in case …. Same results. ** ** Thoughts? ** ** Tony Wyland Messiah College wyl...@messiah.edu ** ** ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.6 Polycom registration problem
Log shows
2012-09-04T20:53:44.422175Z:261:AUTH:ERR:sipxt.vt.messiah.edu:SipRegistrarServer:7f78e2522700:SipRegistrar:Unable
to get credentials for '1...@vt.messiah.edu', realm='vt.messiah.edu',
user='1910/0004f21b526d'
Questions:
1.) Did you run
yum update
after installing ISO to update all rpms, sipx and otherwise?
2.) Does sending server profiles fix issue?
System/Servers/{select server} , Press Send Profiles button
3.) Can you send along sipxconfig.log?
On Wed, Sep 5, 2012 at 1:14 PM, Wyland, Tony wyl...@messiah.edu wrote:
I’m thinking I am missing something stupid here but didn’t figure it out
quite yet. I have been unsuccessful at getting a Polycom 335 to register
with a 4.6 test server. Attached is an interesting snippet from
sipregistrar.log that sounds like some kind of database lookup problem
perhaps. This is the phone attempting to register with line 1 extension
1910. The user was created and assigned to the phone with virtually no
modification from the defaults. The phone device was created by the
auto-provision service. In fact the phone registers just fine before having
a line assigned to it. Here is what was done to get to this point:
· Install new 4.6 server from the ISO followed by a yum update
· Created a new test domain and made this server the 1st server.
DNS entries were created for the test domain and the DNS Advisor reports all
is well.
· Added all of the Telephony Services, Instant Messaging, and Phone
Auto-Provisioning services
· Boot up a Polycom 335 with a cleared config in the test domain.
It found the server, was auto-provisioned, and was registered.
· Changed voicemail dial plan to 4 digits and extension 1900
· Created a user 1910 with password and PIN
· Assigned user 1910 to line 1 on the polycom device and did a Send
Profile
· Phone reboots, comes up with 1910 on the display for Line 1 but is
not able to register
· Changed LOG level for SipRegistrar to INFO level and obtain the
attached snippet during a phone reboot.
· Blew away the server and tried a 2nd round just in case …. Same
results.
Thoughts?
Tony Wyland
Messiah College
wyl...@messiah.edu
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Re: [sipx-users] 4.6 Polycom registration problem
Ah -- the Send Profiles to the server seems to have fixed things up. I saw
that part of the process in the job status screen was to regenerate IMDB. I
can also create other new users and things work as expected now.
I did do a yum update prior to configuring but not prior to the initial setup
script. The GUI shows the following version info: 4.6.0. 2012-09-03EDT17:09:20.
Attached is the sipxconfig.log you requested in case it provides insight on
what may have gone wrong.
While I'm here - we are using Polycom firmware 3.2.6.314. Is it suggested that
we should move up to 3.2.7?
Tony Wyland
Messiah College
wyl...@messiah.edu
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Wednesday, September 05, 2012 1:24 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6 Polycom registration problem
Log shows
2012-09-04T20:53:44.422175Z:261:AUTH:ERR:sipxt.vt.messiah.edu:SipRegistrarServer:7f78e2522700:SipRegistrar:Unable
to get credentials for '1...@vt.messiah.edu', realm='vt.messiah.edu',
user='1910/0004f21b526d'
Questions:
1.) Did you run
yum update
after installing ISO to update all rpms, sipx and otherwise?
2.) Does sending server profiles fix issue?
System/Servers/{select server} , Press Send Profiles button
3.) Can you send along sipxconfig.log?
On Wed, Sep 5, 2012 at 1:14 PM, Wyland, Tony wyl...@messiah.edu wrote:
I'm thinking I am missing something stupid here but didn't figure it out
quite yet. I have been unsuccessful at getting a Polycom 335 to register
with a 4.6 test server. Attached is an interesting snippet from
sipregistrar.log that sounds like some kind of database lookup problem
perhaps. This is the phone attempting to register with line 1 extension
1910. The user was created and assigned to the phone with virtually no
modification from the defaults. The phone device was created by the
auto-provision service. In fact the phone registers just fine before having
a line assigned to it. Here is what was done to get to this point:
* Install new 4.6 server from the ISO followed by a yum update
* Created a new test domain and made this server the 1st server.
DNS entries were created for the test domain and the DNS Advisor reports all
is well.
* Added all of the Telephony Services, Instant Messaging, and Phone
Auto-Provisioning services
* Boot up a Polycom 335 with a cleared config in the test domain.
It found the server, was auto-provisioned, and was registered.
* Changed voicemail dial plan to 4 digits and extension 1900
* Created a user 1910 with password and PIN
* Assigned user 1910 to line 1 on the polycom device and did a Send
Profile
* Phone reboots, comes up with 1910 on the display for Line 1 but is
not able to register
* Changed LOG level for SipRegistrar to INFO level and obtain the
attached snippet during a phone reboot.
* Blew away the server and tried a 2nd round just in case Same
results.
Thoughts?
Tony Wyland
Messiah College
wyl...@messiah.edu
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Re: [sipx-users] 4.6 Polycom registration problem
No reason that I know of to move up. New phones being shipped come with
firmware 3.3 or later and there is a manula downgrade procedure. You can
still register the phones, etc., but they boot up with an error message on
the screen of the phone (at least the ones I had seen) and thought perhaps
it was manifesting itself differently for you whereas the issue was server
send profile needs to be automated somehow (I think) or put a readme/big
fat sticky on something. It is also possible this was fixed in a later
release, but I haven't tested it from a fresh install in a couple of weeks.
On Wed, Sep 5, 2012 at 1:45 PM, Wyland, Tony wyl...@messiah.edu wrote:
Ah -- the Send Profiles to the server seems to have fixed things up. I
saw that part of the process in the job status screen was to regenerate
IMDB. I can also create other new users and things work as expected now.
I did do a yum update prior to configuring but not prior to the initial
setup script. The GUI shows the following version info: 4.6.0.
2012-09-03EDT17:09:20.
Attached is the sipxconfig.log you requested in case it provides insight
on what may have gone wrong.
While I'm here - we are using Polycom firmware 3.2.6.314. Is it suggested
that we should move up to 3.2.7?
Tony Wyland
Messiah College
wyl...@messiah.edu
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Wednesday, September 05, 2012 1:24 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6 Polycom registration problem
Log shows
2012-09-04T20:53:44.422175Z:261:AUTH:ERR:sipxt.vt.messiah.edu:
SipRegistrarServer:7f78e2522700:SipRegistrar:Unable
to get credentials for '1...@vt.messiah.edu', realm='vt.messiah.edu',
user='1910/0004f21b526d'
Questions:
1.) Did you run
yum update
after installing ISO to update all rpms, sipx and otherwise?
2.) Does sending server profiles fix issue?
System/Servers/{select server} , Press Send Profiles button
3.) Can you send along sipxconfig.log?
On Wed, Sep 5, 2012 at 1:14 PM, Wyland, Tony wyl...@messiah.edu wrote:
I'm thinking I am missing something stupid here but didn't figure it out
quite yet. I have been unsuccessful at getting a Polycom 335 to register
with a 4.6 test server. Attached is an interesting snippet from
sipregistrar.log that sounds like some kind of database lookup problem
perhaps. This is the phone attempting to register with line 1 extension
1910. The user was created and assigned to the phone with virtually no
modification from the defaults. The phone device was created by the
auto-provision service. In fact the phone registers just fine before
having
a line assigned to it. Here is what was done to get to this point:
* Install new 4.6 server from the ISO followed by a yum update
* Created a new test domain and made this server the 1st server.
DNS entries were created for the test domain and the DNS Advisor reports
all
is well.
* Added all of the Telephony Services, Instant Messaging, and
Phone
Auto-Provisioning services
* Boot up a Polycom 335 with a cleared config in the test domain.
It found the server, was auto-provisioned, and was registered.
* Changed voicemail dial plan to 4 digits and extension 1900
* Created a user 1910 with password and PIN
* Assigned user 1910 to line 1 on the polycom device and did a
Send
Profile
* Phone reboots, comes up with 1910 on the display for Line 1
but is
not able to register
* Changed LOG level for SipRegistrar to INFO level and obtain the
attached snippet during a phone reboot.
* Blew away the server and tried a 2nd round just in case
Same
results.
Thoughts?
Tony Wyland
Messiah College
wyl...@messiah.edu
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--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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
--
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip:
Re: [sipx-users] TLS handshake fails in 4.6
a little more info... I had been testing by calling the pilot number on the voicemail dialplan (x101). I hadn't gotten as far as turning voicemail from internal to external for a user. When I did enable a user for Exchange, all calls rolled to internal voicemail no matter the settings. Perhaps there's something more fundamentally broken than just the TLS. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS handshake fails in 4.6
On Wed, Sep 5, 2012 at 6:36 PM, John Lightfoot j...@vizhn.com wrote: a little more info... I had been testing by calling the pilot number on the voicemail dialplan (x101). I hadn't gotten as far as turning voicemail from internal to external for a user. When I did enable a user for Exchange, all calls rolled to internal voicemail no matter the settings. Perhaps there's something more fundamentally broken than just the TLS. Well I think it's only related to certs in 4.6 (as you mentioned it works in 4.4) - I see the exception while trying to send invite to ITSP Caused by: javax.net.ssl.SSLHandshakeException: Remote host closed connection during handshake at sun.security.ssl.SSLSocketImpl.readRecord(SSLSocketImpl.java:869) at sun.security.ssl.SSLSocketImpl.performInitialHandshake(SSLSocketImpl.java:1190) at sun.security.ssl.SSLSocketImpl.startHandshake(SSLSocketImpl.java:1217) at sun.security.ssl.SSLSocketImpl.startHandshake(SSLSocketImpl.java:1201) at gov.nist.javax.sip.stack.IOHandler.sendBytes(IOHandler.java:307) at gov.nist.javax.sip.stack.TLSMessageChannel.sendMessage(TLSMessageChannel.java:311) at gov.nist.javax.sip.stack.MessageChannel.sendMessage(MessageChannel.java:255) at gov.nist.javax.sip.stack.SIPTransaction.sendMessage(SIPTransaction.java:745) at gov.nist.javax.sip.stack.SIPClientTransaction.sendMessage(SIPClientTransaction.java:476) at gov.nist.javax.sip.stack.SIPClientTransaction.sendRequest(SIPClientTransaction.java:968) ... 11 more Caused by: java.io.EOFException: SSL peer shut down incorrectly at sun.security.ssl.InputRecord.read(InputRecord.java:352) at sun.security.ssl.SSLSocketImpl.readRecord(SSLSocketImpl.java:850) ... 20 more Can you describe exactly the config steps you took? Thanks George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS handshake fails in 4.6
Sure. For the certs, I installed a StartSSL-issued certificate for my Sipx and two intermediate CAs from Microsoft. I edited the Voicemail dial plan with Exchange Voicemail Server in the type field and the FQDN of the Exchange Online gateway in the address. I added a SIP Trunk gateway with the same Exchange Online address as the dialplan entry and set the transport to TLS and port to 5061. Last, I changed the voicemail server for the user to Microsoft Exchange UM Voicemail Server. I referred earlier to the fact that this last step didn't change the way calls roll to voicemail. Oddly, though, if I dial 8+ext the call does try to go to Exchange rather than to the internal vm. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS handshake fails in 4.6
Because there is no active dial plan entry for 8+ pointing to the exchange system? -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ 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! On Sep 5, 2012 5:45 PM, John Lightfoot j...@vizhn.com wrote: Sure. For the certs, I installed a StartSSL-issued certificate for my Sipx and two intermediate CAs from Microsoft. I edited the Voicemail dial plan with Exchange Voicemail Server in the type field and the FQDN of the Exchange Online gateway in the address. I added a SIP Trunk gateway with the same Exchange Online address as the dialplan entry and set the transport to TLS and port to 5061. Last, I changed the voicemail server for the user to Microsoft Exchange UM Voicemail Server. I referred earlier to the fact that this last step didn't change the way calls roll to voicemail. Oddly, though, if I dial 8+ext the call does try to go to Exchange rather than to the internal vm. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS handshake fails in 4.6
No, the 8 is the default direct-to-voicemail prefix defined no the vm dp. So it's odd when I get the internal voicemail greeting when I timeout on a call to extension 200 but get the TLS attempt when I dial 8200. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
