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SIP Trunk not registering

benjaminvance
Beginner
Beginner

Hello all,

I have a new SIP Trunk I am trying to get to register with the ITSP. I have set the credentials and authentication username and password. When I try to place a call and check the debug I am just sending a Invite. I never see a register being sent. Then I get the 401 Unauthorized. The ITSP says they are seeing nothing coming from me when trying to register. 

Voice service voip

sip
  header-passing
  early-offer forced
  registration passthrough

Debugs:

Sent:
INVITE sip:1574XXXXXXX5@205.196.170.135:5060 SIP/2.0
Via: SIP/2.0/UDP 207.xx.xx.xx:5060;branch=z9hG4bK67759
From: "Collaboration Room" <sip:574XXXXXXX@207.xx.xx.xx>;tag=3BA96E4-D9F
To: <sip:1574XXXXXXX@205.196.170.135>
Date: Tue, 24 Jan 2017 10:20:53 GMT
Call-ID: 87B84949-E17F11E6-80E7DF6E-B03DE5EA@207.xx.xx.xx
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0047750784-0000065536-0000000059-0185860362
User-Agent: Cisco-SIPGateway/IOS-15.6.3.M1
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1485271253
Contact: <sip:574XXXXXXX@207.xx.xx.xx:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-ID: 738dce7f7600a1cae75d8bac7cac6ba0;remote=00000000000000000000000000000000
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 253

Received:
SIP/2.0 100 Trying
Call-ID: 87B84949-E17F11E6-80E7DF6E-B03DE5EA@207.xx.xx.xx
CSeq: 101 INVITE
From: "Collaboration Room" <sip:574XXXXXXX@207.xx.xx.xx>;tag=3BA96E4-D9F
To: <sip:1574XXXXXXX@205.196.170.135>;tag=sip+1+d72100fa+b2471962
Via: SIP/2.0/UDP 207.xx.xx.xx:5060;received=207.xx.xx.xx;branch=z9hG4bK67759
Server: SIP/2.0
Timestamp: 1485271253
Content-Length: 0


*Jan 24 10:20:53 EST: //741/02D89E800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Call-ID: 87B84949-E17F11E6-80E7DF6E-B03DE5EA@207.xx.xx.xx
CSeq: 101 INVITE
From: "Collaboration Room" <sip:574XXXXXXX@207.xx.xx.xx>;tag=3BA96E4-D9F
To: <sip:1574XXXXXXX@205.196.170.135>;tag=sip+1+d72100fa+b2471962
Via: SIP/2.0/UDP 207.xx.xx.xx:5060;received=207.xx.xx.xx;branch=z9hG4bK67759
Content-Length: 0
Supported: resource-priority,siprec

Contact: <sip:205.196.170.135:5060>
WWW-Authenticate: Digest realm="7002619856.siptrunking.appiaservices.com",nonce="d16a4f191953",stale=false,algorithm=MD5,qop="auth"
Server: DC-SIP/2.0
Organization: Metaswitch Networks

13 REPLIES 13

Leszek Wojnarski
Cisco Employee
Cisco Employee

Hi Ben,

Can you share your GW configuration (full if possible), but specifically: sip-ua, and dial-peer sessions would be nice to see.

Leszek

Hi Leszek,

Here is the info you requested.

dial-peer voice 2 voip
 description SIP To APPIA
 destination-pattern 1[2-9]..[2-9]......
 session protocol sipv2
 session target ipv4:205.196.170.135
 incoming called-number .
 voice-class sip bind control source-interface Port-channel1.69
 voice-class sip bind media source-interface Port-channel1.69
 dtmf-relay rtp-nte h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 3 voip
 description SIP To APPIA
 preference 2
 destination-pattern 1[2-9]..[2-9]......
 session protocol sipv2
 session target ipv4:205.196.171.135
 incoming called-number .
 voice-class sip bind control source-interface Port-channel1.69
 voice-class sip bind media source-interface Port-channel1.69
 dtmf-relay rtp-nte h245-alphanumeric
 codec g711ulaw
 no vad

dial-peer voice 5 voip
 description Inbound From MI-CUCM01
 preference 2
 destination-pattern 1[2-9]..[2-9]......
 session protocol sipv2
 session target ipv4:10.X.X.X
 incoming called-number .
 voice-class sip bind control source-interface Port-channel1.195
 voice-class sip bind media source-interface Port-channel1.195
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

sip-ua
 credentials username 7002619856 password 7 xxxxxxxxxxxxxxxx realm siptrunking.appiaservices.com
 authentication username 7002619856 password 7 xxxxxxxxxxxxxxxx
 no remote-party-id
 retry invite 5
 retry register 5
 retry options 10
 timers connect 100
 registrar 1 dns:7002619856.siptrunking1.appiaservices.com expires 3600
 registrar 2 dns:7002619856.siptrunking2.appiaservices.com expires 3600
 registration spike 50
 host-registrar

Can you sen the output from:

sh sip-ua register status

And additionally try to configure:

authentication username 7002619856 password 7 xxxxxxxxxxxxxxxx

on the dial-peer level?

Leszek

Leszek,

I have added the authentication username 7002619856 password 7 xxxxxxxxxxxxxxxx to each of the dial peers. To the ITSP and to CUCM.

VOIP#sh sip-ua register status
--------------------- Registrar-Index  1 ---------------------

Line                             peer       expires(sec) reg survival P-Associ-URI
================================ ========== ============ === ======== ============
7002617001                       -1         0            no  normal

--------------------- Registrar-Index  2 ---------------------

Line                             peer       expires(sec) reg survival P-Associ-URI
================================ ========== ============ === ======== ============
7002617001                       -1         57           no  normal

When I place a test call I am getting "your can not be completed" or a busy signal.