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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.

Hi Ben

As per my understand of what's expected from telco (I don;t know what's been agreed with them), is the call flow similar to:

http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-aaa.html

Figure 3. UAC-to-UAS Call Flow with INVITE Message

In your case CUBE doesn't send second INVITE or we cannot see it.

Can you send the debugs:
debug voice ccapi inout and debuc ccsip messages from the time when you try to make a call.

You mentioned that you have two types of failures that are happening so it would be good to have debugs from both.

Leszek

Leszek,

I am still looking over the SIP aaa document you sent. I think this might help but I have not had time to really go over it. I have attached the debug out put from the ccsip messages and the voice ccapi inout. With the two types of failures I get the router out put is the same I just get different messages from the phone. Either "your call can't be completed as dialed" or a busy signal. The one thing I am seeing is that I send the Invite in the sip message but seem to be sending an Register request. Also should the SIP trunk always be registered with the carrier or just when a call is being placed?

Thanks for your help.

Ben

Hi Ben,

From debugs I don't see anything wrong with the Authentication setup itself, it looks more like issue with the format of the number or something like this as telco is sending: 480 and 404, which would mean that they don;t have this destination number available.

So I'd suggest checking with telco to verify what number to the expect (how many digits and what prefix). Or thye might be able to tell right away once you show they the debugs from this of why they send 480 and 404, as from our perspective we only receive it and cannot really tell why, they should tell.

Jan 27 15:33:16 EST: //2438/DDA892000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 480 Temporarily Unavailable
Call-ID: AA655189-E40611E6-8285DF6E-B03DE5EA@207.32.250.245
CSeq: 102 INVITE
From: "Collaboration Room" <sip:574XXXXXXX@207.32.250.245>;tag=143CB468-1459
To: <sip:1574XXXXXXX@205.X.X.X>;tag=sip+1+c8c80055+2ea3b9ff
Via: SIP/2.0/UDP 207.32.250.245:5060;received=207.32.250.245;branch=z9hG4bK12421D2
Content-Length: 0
Supported: resource-priority,siprec
Contact: <sip:205.X.X.X:5060>
Server: DC-SIP/2.0
Organization: Metaswitch Networks

Jan 27 15:33:16 EST: //2439/DDA892000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.X.X.X:5060;branch=z9hG4bK1251D21
From: "Collaboration Room" <sip:574XXXXXXX@172.X.X.X>;tag=143CB6E8-1A9E
To: <sip:1574XXXXXXX@10.X.X.X1>;tag=203~c2551668-bbbd-4dfe-a185-df4baa6ae3ae-47421692
Date: Fri, 27 Jan 2017 20:33:34 GMT
Call-ID: AAC6F838-E40611E6-8287DF6E-B03DE5EA@172.X.X.X
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Server: Cisco-CUCM11.5
Session-ID: 00000000000000000000000000000000;remote=738dce7f7600a1cae75d8bac7cac6ba0
Content-Length: 0

HTH,

Leszek

Leszek,

I have talked with the Carrier and they confirmed that they only need 10 digits so I have made the adjustments on my dial peers for that. But with that I am still seeing the same issue. They did say they could see the calls hitting their gate way and they are rejecting them. They do not see a register request coming from us. As I do not see us sending a register request to them in any Invite message. Do know of a way to debug why register requests are not being sent?

SIP/2.0 480 Temporarily Not Available
Via: SIP/2.0/TCP 10.1.20.11:5060;branch=z9hG4bK12a7fae4809
From: "Collaboration Room" <sip:7002617001@10.X.X.X>;tag=226~c2551668-bbbd-4dfe-a185-df4baa6ae3ae-47421764
To: <sip:574XXXXXXX@172.X.X.X>;tag=224A3720-85B
Date: Mon, 30 Jan 2017 09:02:41 GMT
Call-ID: cd253100-88f14793-83-b14010a@10.1.20.11
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.6.3.M1
Reason: Q.850;cause=20
Session-ID: 00000000000000000000000000000000;remote=738dce7f7600a1cae75d8bac7cac6ba0
Content-Length: 0

Thanks,

Ben

Configuration seems to be fine from what I could see and you CUBE should ate least attempt to Register to TELCO. The only thing that still can be checked it if your CUBE can ping/resolve both registrar servers:

7002619856.siptrunking1.appiaservices.com

you might want to try to define this via IP if that's possible in your case:

registrar ipv4:10.10.10.10 expires 180

So basically your CUBE should start sending REGISTER messages just after  "registrar" command is added under sip-ua.

So the action plan I'd suggest:

1. Try to ping your servers via DNS.

2. If you can ping then remove registrar commands from sip-ua.

3. Enabled debugs ccsip messages.

4. Add registrar commands at this point your CUBE should start sending REGISTER messages. It will try couple of times and the stops trying if this fails.

Leszek

Leszek,

So the issue we have is with VRF routing. Both the inside and outside interfaces are in the same VRF but for what ever reason any thing that happens internal to the CUBE is not being passed to the VRF interfaces. When we would do a packet capture we could see things going to the CUBE from the inside and see things leaving the CUBE from the outside interface but nothing in between. So I was able to get the CUBE to register with the ITSP by removing the VRF router then it register right up. We are trying to use this CUBE for multiple customers using VRF's. So each customer will have their own VRF to a SIP trunk and a CUCM cluster. We are working on trying to find out what we are missing now with the VRF routing. Thanks for all your support on this issue.

Thanks,

Ben

Hi,

Couple of things which aren't correct here:

1. Your realm is pointing to siptrunking.appiaservices.com while your registrar points to different domain names (siptrunking1.appiaservices.com & siptrunking2.appiaservices.com). Is this correct? I don't think so.

2. You dial-peer are pointing are pointing to IP address rather than domain name of the registrar. Where you got the IP address from? Are you sure that domain name points to the right IP address.

3. I just to create sip-server under your sip-ua and point your dial-peers to sip-server

4. Suggest you configure authentication realm as well.

Hi Mohammed,

The realm is correct the siptrunking.appiaserices.com is an SVR record which points to siptrunking1 and siptrunking2.appiaservice.com. The IP address are the address that the SIP trunk provider gave use to point to. I was able to get this working by removing the VRF routing that was causing the issue.

Thanks,

Ben

Good that you fixed. I wasn't aware that you are using VRFs. You have fixed it by enabling vrf aware voice instead of removing the vrfs. The command is 'voice vrf ###'. Before 15.5.x one vrf instance was supported.  In later releases 5 vrfs are supported.  check out the release notes to get the exact IOS

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