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Need Help setting up Sip Trunk

jacobpslagle
Level 1
Level 1

Hello,

 

I have been spending hours the past few days trying to get my SIP trunk configured for my deployment of CUCME. When I try to make an outbound call all I get is that busy signal and unknown number. When I try to call in all I get is "This number you are trying to reach is not a valid number". I have no idea what's going on I am still new to CME but everything else is working fine. I have attached my error log and router configuration. Hopefully I get this this matter resolved.

 

Jake

29 Replies 29

Dennis Mink
VIP Alumni
VIP Alumni

looking at the log you are getting the following invite:

 

Received:
INVITE sip:6011972595083957@10.1.0.5 SIP/2.0
To: 6011972595083957<sip:6011972595083957@10.1.0.5>
From: 1002<sip:1002@10.1.0.5>;tag=0ec07a6d
Via: SIP/2.0/UDP 46.166.151.142:5071;branch=z9hG4bK-0316ed9ec7fc5f0bff9932315361dd60;rport
Call-ID: 0316ed9ec7fc5f0bff9932315361dd60
CSeq: 1 INVITE
Contact: <sip:1002@46.166.151.142:5071>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE
User-Agent: sipcli/v1.8
Content-Type: application/sdp
Content-Length: 285

 

you are trying to call 6011972595083957 and there is no dialpeer for it. I am guussing you need to strip of the leading 60

 

cheers

Please remember to rate useful posts, by clicking on the stars below.

That number does not look familiar I am not sure if it is supposed to and any idea what that dialpeer may look like? 

Hi @jacobpslagle,

 

 

do you try this:

R1#show dialplan number "CALLED NUMBER"

in order to check which dial-peer your call could take.

 

You have got 2 dial-peers, 1 & 2.

None destination-pattern match with the number called: "6011972595083957".

 

 

 

 

Best regards
.ı|ı.ı|ı. If This Helps, Please Rate .ı|ı.ı|ı.

R0g22
Cisco Employee
Cisco Employee
A couple of things -

1. Does your ITSP require the SIP trunk to be registered ?
2. From the logs I see the calls being rejected with CV=21. Does the router trust list have the IP of your ITSP who is the originator of calls for inbound calls ? If you already have dial-peers with a session target of your ITSP you should not need it.

To add to this, the 401 message is giving a realm of "sip.flowroute.com" as seen here:

 

May 24 01:41:19.147: //43/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 71.58.129.156:61499;rport=50340;branch=z9hG4bK1C146F
From: <sip:12407506433@us-east-va.sip.flowroute.com>;tag=158348-10C1
To: <sip:12407506433@us-east-va.sip.flowroute.com>;tag=870a5b262384e4f9f82f59836d699db5.c945
Call-ID: 7195137A-5E2011E8-8002B6FF-3C4B5985
CSeq: 13 REGISTER
WWW-Authenticate: Digest realm="sip.flowroute.com", nonce="WwYZa1sGGD/RoIqViquUZTNgEDObmgWs", qop="auth"
Content-Length: 0

 

However, your authentication line has "us-east-va.sip.flowroute.com" for the realm:


sip-ua
authentication username 27574689 password 7 06110E29615A5F0156413F260D realm us-east-va.sip.flowroute.com

 

Can you try changing the realm on your authentication line from "us-east-va.sip.flowroute.com" to "sip.flowroute.com?"  This should allow your CME to reply to the 401 with a new REGISTER message containing the username/password.

@gmgarrian

 

Changing the realm in my sip-ua authentication line from "us-east-va.sip.flowroute.com" to "sip.flowroute.com worked I can finally make outbound calls. It does not make any sense because the guide that I followed told me to replace all instances of sip.flowroute.com to us-east-va.sip.flowroute.com if I was using their POP feature which in this case I am. Here is a link to their guide:  

 

https://support.flowroute.com/SIP_Trunking_and_Voice/PBX__Configuration_Guides/Cisco/Configure_Cisco_Unified_Communications_Manager

 

So I can make outbound calls but no inbound ones. All I get is "You call cannot be completed at this time".

 

I just want the number to forward straight to the phone.

 

DIDS are: 

e-phone 1 has 12407506433 Type 7962

e-phone 2 has14129604388 - Type 7975

 

An updated router config has been linked but I feel like this has something to do with the dial peers, I am not sure. According to the documentation, I should only need those 2. I really appreciate all the help from everybody. 

Documentation and reality are too often very different and we need to go by the realm in the 401 SIP message.  This type of mix up happens very often.

 

Your dial-peers look fine but they are redundant.  I would add "incoming called-number ." to one of them so it will match for inbound calls from your provider.

 

The first thing to verify for inbound calls is if the SIP invite is making it to your CME.  Can you get these debugs for a test call:

 

debug ccsip message

debug voip ccapi inout

 

Your phones will create virtual dial peers when they register.  Since outbound calls work I'm assuming they are registered and the virtual peers are there.  You can verify with:

 

show dial-peer voice summary

 

The debugs will tell us where to focus next.

what would I want to add for incoming called-number?

 

Also for some reason now I can't send out calls.

 

I attached 2 logs one for inbound and one for outbound.

You may need to contact your provider on this one.  Now they're sending the 401 Unauthorized even after the new REGISTER message containing the username/password is sent:

 

May 24 14:38:43.225: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:us-east-va.sip.flowroute.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.5:5060;branch=z9hG4bKC1C55
From: <sip:1001@us-east-va.sip.flowroute.com>;tag=8DD474-2348
To: <sip:1001@us-east-va.sip.flowroute.com>
Date: Thu, 24 May 2018 14:38:43 GMT
Call-ID: 58D258F7-5E8D11E8-802AB0E7-486F6881
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M3
Max-Forwards: 70
Timestamp: 1527172723
CSeq: 19 REGISTER
Contact: <sip:1001@10.1.0.5:5060>
Expires: 3600
Authorization: Digest username="27574689",realm="sip.flowroute.com",uri="sip:us-east-va.sip.flowroute.com:5060",response="bc653a958b764a64721bda7f1c36f28e",nonce="WwbPn1sGznOjIYH9FJ7mFDXqhJ4j2gUZ",cnonce="F5E21009",qop=auth,algorithm=md5,nc=00000001
Content-Length: 0


May 24 14:38:43.337: //172/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 71.58.129.156:62338;rport=55858;branch=z9hG4bKC0283
From: <sip:12407506433@us-east-va.sip.flowroute.com>;tag=8DD464-D04
To: <sip:12407506433@us-east-va.sip.flowroute.com>;tag=870a5b262384e4f9f82f59836d699db5.0136
Call-ID: 58D258F7-5E8D11E8-802BB0E7-486F6881
CSeq: 19 REGISTER
WWW-Authenticate: Digest realm="sip.flowroute.com", nonce="WwbPn1sGznOjIYH9FJ7mFDXqhJ4j2gUZ", qop="auth"
Content-Length: 0

Can you verify the username a password are correct?   

Username and Password are correct and would that be my SIP Trunk provider?

Yes, your SIP provider is not accepting the registration message.  Once they tell us why we can make the appropriate changes on our end if any are needed.  

 

Did anything change since this was working after making the change to the authentication parameters?

Alright I am going to call them now and I don't believe so, here is my current router config and here is the sip log from the past 5 minutes.

Alright so I called them and it looks like they are no accepting the registration and they have to do some more digging into the matter. I will keep this ticket updated. 

Sounds good.  I think we're in the home stretch!