Bringing this back to the mailing list. It seems Roman has the same issue.
On 08/17/2011 12:06 AM, cyril constantin wrote:
I have identified the issue,
If I create a Permissions named "Avaya Asterisk Extensions
<https://ps0sipx01.webcti.local:8443/sipxconfig/permission/ListPermissions,editRowLink.sdirect?sp=9&state:permission/ListPermissions=BrO0ABXcnAAAAAQEADiRjb21tb24kQm9yZGVyABBpbml0aWFsU2Vzc2lvbklkdAANMTZlZXJ0d2lnNzRpOQ%3D%3D>"
and then that I go to Dial Plan and then I create an "Avaya
extensions" rule, if I use permission "Avaya Asterisk extensions" my
calls will not work, but if I use
Permission "Local Dialing" or something else it will works.
I have checked users permissions and I can see that "Avaya Asterisk
Extensions" is checked.
It looks that permissions are not correctly authorized to users if
it's a custom Permissions and not a System Permissions
Let me know if you can reproduce my case.
Regards.
2011/8/16 cyril constantin <[email protected]
<mailto:[email protected]>>
I have triple check my permission and remove everything to create
it again but it doesn't work, I really believe it's related to the
new release because it was working before the upgrade.
I'll continue to investigate.
Thanks for your feedback.
Regards.
2011/8/16 Joegen Baclor <[email protected] <mailto:[email protected]>>
You have a permission issue. This is more of a config thing.
Please post this log to the mailing list. Perhaps the config
guys would butt in.
SIP/2.0 403 Requires perm_7\r
From: \"Cyril CONSTANTIN\"<sip:[email protected]>;tag=7448281f\r
To: <sip:[email protected]>;tag=RmfYZM\r
Call-Id: ZGVhZjllNGI1YzY0MmUxODg2MGVlYWIxMGIxODE4MGU.\r
Cseq: 2 INVITE\r
Via: SIP/2.0/UDP
10.147.113.221;branch=z9hG4bK-XX-002f0gdiURKEgD_XyifXFGVjYQ~45Lz98idGtRFn80`e636vg\r
Via: SIP/2.0/TCP
10.147.116.81:12686;branch=z9hG4bK-d8754z-469413e29a96945a-1---d8754z-;rport=50838\r
Record-Route: <sip:10.147.113.221:5060;lr;sipXecs-CallDest=CUST>\r
Server: sipXecs/4.4.0 sipXecs/sipXproxy (Linux)\r
Date: Tue, 16 Aug 2011 13:49:20 GMT\r
Content-Length: 0\r
On 08/16/2011 09:55 PM, cyril constantin wrote:
Joegen,
Please find in attach my test call? I'm trying to call number
0625081954 <tel:0625081954> from my sip
peers 46000 registered on sipxecs, the call normally goes
trough the unmanaged gateway Asterisk it's adress is
sip.118218.fr <http://sip.118218.fr>.
Logs are set to Debug for Sip Proxy.
Please let me know for any questions.
It was working like
Best Regards.
211/8/16 Joegen Baclor <[email protected]
<mailto:[email protected]>>
Only debug level would make sense. Make sure you send a
test call and send me the call information
On 08/16/2011 09:36 PM, cyril constantin wrote:
In which log level do you need it ?
2011/8/16 Joegen Baclor <[email protected]
<mailto:[email protected]>>
That warning is not an error. the significant log
is the proxy log, not the registrar.
On 08/16/2011 08:59 PM, cyril constantin wrote:
I forgot to precise that my unmanaged gateway
behing is Asterisk
2011/8/16 cyril constantin
<[email protected]
<mailto:[email protected]>>
Below traces from sipregistrar.log:
"2011-08-16T12:55:40.642212Z":60:SIP:NOTICE:ps0sipx01.webcti.local:SipRedirectServer-17:4058B940:SipRegistrar:"ContactList::add():
[140-FALLBACK] SipRedirectorFallback added
contact for 'sip:0625081954
<tel:0625081954>@webcti.local;transport=tcp':\n
'<sip:0625081954
<tel:0625081954>@sip.118218.fr:5060;transport=tcp;sipxecs-lineid=3;sipXecs-CallDest=CUST?expires=7200>;q=0.9'
(contact index 0)"
"2011-08-16T12:55:40.642373Z":61:SIP:NOTICE:ps0sipx01.webcti.local:SipRedirectServer-17:4058B940:SipRegistrar:"ContactList::set():
[999-AUTHROUTER] SipRedirectorAuthRouter
modified contact index 0 for 'sip:0625081954
<tel:0625081954>@webcti.local;transport=tcp':\n
was: '<sip:0625081954
<tel:0625081954>@sip.118218.fr:5060;transport=tcp;sipxecs-lineid=3;sipXecs-CallDest=CUST?expires=7200>;q=0.9'\n
now is: '<sip:0625081954
<tel:0625081954>@sip.118218.fr:5060;transport=tcp;sipxecs-lineid=3;sipXecs-CallDest=CUST?expires=7200&ROUTE=%3Csip%3Awebcti.local%3Blr%3E>;q=0.9'"
"2011-08-16T12:55:50.392637Z":62:HTTP:WARNING:ps0sipx01.webcti.local:HttpConnection-51:42FD8940:SipRegistrar:"HttpConnection::run
read 0 bytes, indicating peer shut down"
2011/8/16 cyril constantin
<[email protected]
<mailto:[email protected]>>
Hi,
I'm facing the same issue than roman, I
can't use my Unmanaged gateway sinc I have
upgraded this morning, please find below
logs from sipXproxy:
"2011-08-16T12:48:20.698773Z":23:SIP:ERR:ps0sipx01.webcti.local:SipRouter-15:41B0F940:SipXProxy:"Url::parseString
no valid host found at char 0 in '',
uriForm = name-addr"
"2011-08-16T12:48:20.703172Z":24:SIP:WARNING:ps0sipx01.webcti.local:SipClientTcp-19:41129940:SipXProxy:"SipUserAgent::dispatch
received response without transaction"
"2011-08-16T12:48:20.723828Z":25:AUTH:WARNING:ps0sipx01.webcti.local:SipRouter-15:41B0F940:SipXProxy:"EnforceAuthRules[400_authrules]::authorizeAndModify
call
'MjlkZGE5YTljNzcyMzYyZjk0MTBlMzQ3MDI0M2YzYjQ.'
requires 'perm_6'"
"2011-08-16T12:48:20.811714Z":26:SIP:ERR:ps0sipx01.webcti.local:SipUserAgent-2:4189E940:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no
matching transaction"
Any idea?
Best Regards.
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