CUCME - SIP phone calls can't match incoming phone number

Pawel Lenart · ‎10-18-2012

Hi all,

I have a problem with new SIP trunk. I configured SIP trunk but for some reason CUCME is taking registration username and it's trying to match called number with this username instead of maching phone number with invite messages. My debug (debug ccsip messages, debug ccsip calls)

If anyone will need part of my configuration please let me know.

My SIP trunk:

username: test (character based username)

Router internal IP address: 192.168.1.1

Proxy IP addresses: 222.222.222.222 & 111.111.111.111

Number connected to SIP trunk (called number): 01111111111 (441111111111)

My mobile number (calling number): 07777777777

Phone numbers connected to SIP trunk: 0

My debug:

Oct 18 14:38:31.177: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:test@192.168.1.1:5060 SIP/2.0

From: <sip:07777777777@pstn1>;tag=8DBE20DC-E2C

To: <sip:441111111111@lon-1.e164.org.uk>

Call-ID: 078a0547a376b21747d6d70b81634319fd3a5a7@222.222.222.222

CSeq: 101 INVITE

Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK-a83902-911eafd8-7a3ad46

Content-Type: application/sdp

Contact: <sip:07777777777@222.222.222.222:5060;nt_end_pt=YM0+~K.NCzzyfw360~QJ1m0dPr75a160~KIi60~EbtGkm~WncUzV-1_.~LnirPQGRelF!c~PMhI-A~Xn7ZQzN~LEtb-3BBDX1.6GOd5t-V3E6*6o1UQPdtm.F4~Gzqo13.5Pnq50dP0;nt_server_host=222.222.222.222:5060>

User-Agent: MSSGW

Expires: 180

Max-Forwards: 13

Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,timer,resource-priority,replaces

Remote-Party-ID: <sip:07777777777@213.166.5.160>;screen=yes;screen-ind=0;party=calling;counter=0;privacy=off

Allow: INVITE,BYE,CANCEL,ACK

x-nt-corr-id: e68a0547a3ed1851747d6d709cc3ee63d31833b6@222.222.222.222

x-nt-location: 4057

Allow-Events: telephone-event

Date: Thu, 18 Oct 2012 14:38:31 GMT

Timestamp: 1350571111

Privacy: none

Content-Disposition: session;handling=required

x-nt-service: brdplayed=yes

Min-SE: 1800

Content-Length: 417

v=0

o=CiscoSystemsSIP-GW-UserAgent 8067 9285 IN IP4 111.111.111.111

s=SIP Call

e=unknown@invalid.net

t=0 0

m=audio 48606 RTP/AVP 8 18 4 3 98 0 101

c=IN IP4 111.111.111.111

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:4 G723/8000

a=fmtp:4 bitrate=6.3;annexa=no

a=rtpmap:3 GSM/8000

a=rtpmap:98 G726-32/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

Oct 18 14:38:31.185: //24718/525930A88869/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK-a83902-911eafd8-7a3ad46

From: <sip:07777777777@pstn1>;tag=8DBE20DC-E2C

To: <sip:441111111111@lon-1.e164.org.uk>

Date: Thu, 18 Oct 2012 14:38:31 GMT

Call-ID: 078a0547a376b21747d6d70b81634319fd3a5a7@222.222.222.222

Timestamp: 1350571111

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2.4.M1

Content-Length: 0

Oct 18 14:38:31.185: //24718/525930A88869/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK-a83902-911eafd8-7a3ad46

From: <sip:07777777777@pstn1>;tag=8DBE20DC-E2C

To: <sip:441111111111@lon-1.e164.org.uk>;tag=76FC11F8-58B

Date: Thu, 18 Oct 2012 14:38:31 GMT

Call-ID: 078a0547a376b21747d6d70b81634319fd3a5a7@222.222.222.222

Timestamp: 1350571111

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2.4.M1

Reason: Q.850;cause=28

Content-Length: 0

Oct 18 14:38:31.201: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:test@192.168.1.1:5060 SIP/2.0

From: <sip:07777777777@pstn1>;tag=8DBE20DC-E2C

To: <sip:441111111111@lon-1.e164.org.uk>;tag=76FC11F8-58B

Call-ID: 078a0547a376b21747d6d70b81634319fd3a5a7@222.222.222.222

CSeq: 101 ACK

Via: SIP/2.0/UDP 222.222.222.222:5060;rport=53434;branch=z9hG4bK-a83902-911eafd8-7a3ad46

Max-Forwards: 70

Content-Length: 0

Oct 18 14:38:31.201: //24718/525930A88869/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x0x33BCF798

State of The Call : STATE_DEAD

TCP Sockets Used : NO

Calling Number : 07777777777

Called Number : test

Source IP Address (Sig ): 192.168.1.1

Destn SIP Req Addr:Port : 222.222.222.222:5060

Destn SIP Resp Addr:Port : 222.222.222.222:5060

Destination Name : 222.222.222.222

Oct 18 14:38:31.201: //24718/525930A88869/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream : 1

Negotiated Codec : g711alaw

Negotiated Codec Bytes : 160

Nego. Codec payload : 8 (tx), 8 (rx)

Negotiated Dtmf-relay : 6

Dtmf-relay Payload : 101 (tx), 101 (rx)

Source IP Address (Media): 192.168.1.1

Source IP Port (Media): 16610

Destn IP Address (Media): 111.111.111.111

Destn IP Port (Media): 48606

Orig Destn IP Address:Port (Media): [ - ]:0

Oct 18 14:38:31.201: //24718/525930A88869/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC) : 28

Disconnect Cause (SIP) : 484

ADAM CRISP · ‎10-22-2012

This is called the Request URI

INVITE sip:test@192.168.1.1:5060 SIP/2.0

and this is the only header your cisco will use to route calls.

This is called the to header:

To: <>441111111111@lon-1.e164.org.uk>

Cisco don't support the routing of calls using the To header

When your phone system "registers" with the service provider it is registering a "contact". The contact forms part of the AOR (Address of Record) and classic use of the registration process is intended to tell the SP what "contacts" are available on site.

So it's likely that your router is registering the contact "test" where, it's probably better if it registers your DDI/DID.

There are two things I can think to try

1. get the router to register your DDI/DDI by using the credentials command under SIP-UA

2. Ask your Service Provider. Since it's unreasonable and not desirable to go through the pain of registering every DDI, it's likely that your SP has a request URI rewrite tool, where a single SIP registation (test in your case) can be used to identify the whole site. They should be able to re-write the request URI in the form your router needs:

i.e.

INVITE sip:441111111111@192.168.1.1:5060 SIP/2.0

I hope this helps.

Adam

Pawel Lenart · ‎10-23-2012

Thanks Adam. Very helpful.