[cisco-voip] CME w/ SIP Trunk

[Ed Leatherman]

Following up for posterity..

Still fighting this one but a small bit of progress. So CME was trying to
register both the ephone-dn extension and the E164 expanded number of the
extension (which was the correct one) - so SP was putting us in timeout for
trying to register invalid numbers. I figured out how to stop that with the
number xxx no-reg command in each ephone-dn.

Also, I had to re-write some headers in the REGISTER requests
(to/from/request-uri and Authorization fields) with sip profiles.

Also - as brian suggested, they told me the wrong username, should have not
had dashes in it.

After all that, still getting back 403 Authentication Failed - but its at
least cleaned up and I'm not getting put in timeout.

onward..



On Tue, Jun 23, 2015 at 6:06 PM, Ed Leatherman <ealeatherman at gmail.com>
wrote:

> Didnt seem to help but thats a good thought. Slogging it out with SP
> tomorrow again. I'm really puzzled about the SIP debugs.. its like i'm not
> getting all my SIP messages in there
>
> On Tue, Jun 23, 2015 at 3:29 PM, Brian Meade <bmeade90 at vt.edu> wrote:
>
>> I'd try the username without the dashes first.
>>
>> On Tue, Jun 23, 2015 at 3:26 PM, Ed Leatherman <ealeatherman at gmail.com>
>> wrote:
>>
>>> I did a packet cap and we are sending the SIP REGISTER, but its not
>>> showing up in sip debug?? really weird. anywhere I'm not binding SIP to my
>>> loopback address, i'm not getting SIP debugs for.
>>>
>>> So I am getting 403 back from SP after all, gonna double check
>>> username/passwords
>>>
>>> On Tue, Jun 23, 2015 at 3:16 PM, Brian Meade <bmeade90 at vt.edu> wrote:
>>>
>>>> How about connecting via telnet over 5060?  You may be having a TCP
>>>> issue which is why you never see the Register sent.
>>>>
>>>> On Tue, Jun 23, 2015 at 3:09 PM, Ed Leatherman <ealeatherman at gmail.com>
>>>> wrote:
>>>>
>>>>> Brian
>>>>> msu-tmp-access#sho sip-ua register status
>>>>> Line                             peer       expires(sec) reg survival
>>>>> P-Associ-URI
>>>>> ================================ ========== ============ === ========
>>>>> ============
>>>>> 20311                            20001      43           no  normal
>>>>> 20312                            20003      43           no  normal
>>>>> 20313                            20005      43           no  normal
>>>>> 20314                            20007      43           no  normal
>>>>> .. etc .. all no
>>>>>
>>>>> I can ping the sip-server from router so it appears to be able to
>>>>> resolve the name OK.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jun 23, 2015 at 2:48 PM, Brian Meade <bmeade90 at vt.edu> wrote:
>>>>>
>>>>>> What do you see for "show sip-ua register status"?  Are you sure the
>>>>>> gateway can resolve the sip-server via DNS?
>>>>>>
>>>>>> On Tue, Jun 23, 2015 at 2:32 PM, Ed Leatherman <
>>>>>> ealeatherman at gmail.com> wrote:
>>>>>>
>>>>>>> Hello!
>>>>>>>
>>>>>>> I'm trying to get a SIP trunk out to a regional SP (Lumos)
>>>>>>> configured. I need to get CME setup to register numbers with their sip
>>>>>>> proxy, but the registration is not happening and i'm not seeing any
>>>>>>> register messages debugs from debug ccsip messages to troubleshoot from. So
>>>>>>> I think maybe CME isn't trying? What should trigger CME to try and register
>>>>>>> these numbers?
>>>>>>>
>>>>>>> My config looks like this (some ephones/ephone-dns up and
>>>>>>> registered) - authentication credentials were provided from Lumos. IOS
>>>>>>> 15.4(3)M2
>>>>>>>
>>>>>>> msu-tmp-access#sh run | s sip-ua
>>>>>>> sip-ua
>>>>>>>  credentials username 304-929-0300 password 7 blah realm
>>>>>>> sbc.ia.ntelos.net
>>>>>>>  authentication username 304-929-0300 password 7 blah
>>>>>>>  retry register 10
>>>>>>>  registrar dns:sbc.ia.ntelos.net:5060 expires 120
>>>>>>>  sip-server dns:sbc.ia.ntelos.net:5060
>>>>>>> !
>>>>>>> msu-tmp-access#sho run | s voice service
>>>>>>> voice service voip
>>>>>>>  ip address trusted list
>>>>>>>   ipv4 216.12.114.195
>>>>>>>  address-hiding
>>>>>>>  allow-connections sip to sip
>>>>>>>  fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback
>>>>>>> none
>>>>>>>  sip
>>>>>>>   bind control source-interface GigabitEthernet0/2
>>>>>>>   bind media source-interface GigabitEthernet0/2
>>>>>>>   registrar server
>>>>>>>   options-ping 60
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Ed Leatherman
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> cisco-voip mailing list
>>>>>>> cisco-voip at puck.nether.net
>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Ed Leatherman
>>>>>
>>>>
>>>>
>>>
>>>
>>> --
>>> Ed Leatherman
>>>
>>
>>
>
>
> --
> Ed Leatherman
>



-- 
Ed Leatherman
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20150624/a96eb201/attachment.html>