10-18-2012 07:59 AM - edited 03-16-2019 01:45 PM
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
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
10-22-2012 02:39 PM
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
10-23-2012 04:58 AM
Thanks Adam. Very helpful.
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