CME 7.0 and Nokia E5 with SIP Client - cannot receive calls

Unanswered Question
Nov 12th, 2010


i have running a working system with CME 7.0 on a 2811 Router. IP Phones 7961,7921 both SCCP Clients, and a Nokia E65, E5 both SIP Clients.
The E65 are working without problems, but the E5 with VoiPSettings 3.x installed cannot receive calls - neither from E65 and SCCP Phones?

has someone any idea, where i can try to find the error? It seems that the VoipSettingsTool 3.x from Nokia is not working well...


I have this problem too.
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LBS-BZ-HI Fri, 11/12/2010 - 04:14


i think the register process is ok, the only difference between the E65 and E5 is following:

E5: Contact: ;expires=3600   (Here contact is a strange value)

E65: Contact: ;expires=3600

is that normal?


Felipe Garrido Mon, 11/15/2010 - 08:25

Can you collect the following debugs for a call made to the E5?

debug voip ccapi inout

debug ccsip message

Also, can you collect the following command output.

show dial-peer voice summary


Felipe Garrido Mon, 11/15/2010 - 08:45


the debugs indicate that the E5 itself is rejecting the call.

SIP/2.0 404 Not Found

What extension does it show on the phone itself? Does it show 781 or something else?


LBS-BZ-HI Wed, 11/17/2010 - 02:30

it shows me following:




sip:; transport= UDP


i didn't found more info.

Felipe Garrido Wed, 11/17/2010 - 06:49

what client are you using on the Nokia phones that supports SIP registration? I have a E65 here in the lab that I can play with but it doesn't have the SIP client.


LBS-BZ-HI Wed, 11/17/2010 - 23:19

i have Nokia E65 with "SIP Voip 1.0 Settings"- they are working. but the nokia E5 installed with "SIP Voip 3.x Settings" isn't working... i think the SIP client from nokia has a bug...

Felipe Garrido Thu, 11/18/2010 - 07:02

Can you try downgrading to version 1.0 on the E5? I'll check to see if I can upgrade the E65 here in our labs.

LBS-BZ-HI Fri, 11/19/2010 - 03:11

i cannot install the old version on the E5..

. but i found following in a forum:


When VoIP3.x registers a user, it writes a hash in the contact URI, i.e. instead of

To invite a user, the server has to put this hash in the request-URI


sounds not good...

jsalas Tue, 11/30/2010 - 15:59

Try this to change the directory number of the Nokia to the hash in the SIP req uri  header:

voice service voip
allow-connections sip to sip
  bind control source-interface XXX
  bind media source-interface XXX

  registrar server
  sip-profiles 1
voice class sip-profiles 1
request INVITE sip-header SIP-Req-URI modify "sip:752@(.*) SIP/2.0" "sip:3HkqSha_SBZaN-Mrfjb4@\1 SIP/2.0"


voice register global
mode cme
source-address XXXX port 9090
no outbound-proxy
no privacy
max-dn 20
max-pool 20
authenticate realm XXXX

create profile sync 0554118347449519
voice register dn  1
number 752

name E52
voice register dn  2
number 771
name E71

To capture the hash, do a debug ccsip messages and look at the REGISTER message.


mroggia Tue, 05/24/2011 - 04:07

I'm in the same exact  situation, voip setting 3.x give a chance just to place and cannot receive.

i'm quite sure that is something on nokia device

mroggia Tue, 05/24/2011 - 05:13

In my case, i miss ringing sip response from E5

*May 24 12:10:28.730: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP;branch=z9hG4bK681F5
To: ;tag=2k9lum34hphc73eb43l8
From: "CED" ;tag=19D5D654-383
Call-ID: A60943D7-853511E0-A849EB83-64D64ED2@
CSeq: 101 INVITE
Content-Length: 0

while iphone for example that work also receiving the answer in ringing is:

*May 24 12:11:54.178: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP;received=;branch=z9hG4bK69B08
Call-ID: D8D2F8A1-853511E0-A85FEB83-64D64ED2@
From: "CED" ;tag=19D7232C-796
To: ;tag=jurshl3rxxg30qf02oztazib9x5tsqcg
CSeq: 101 INVITE
Contact: "4005"
Content-Length:  0

mroggia Wed, 05/25/2011 - 05:20

I found the solution...

every nokia present the SIP INVITE changing the telephone number with an ASCII like this : ZVW-kntKmiPbSiXyX9zP


it's enough create a voice class sip-profile 1

request INVITE sip-header SIP-Req-URI modify "sip:4006@(.*) SIP/2.0" "sip:ZVW-kntKmiPbSiXyX9zP@\1 SIP/2.0"

in my case 4006 is my telephone number.... with the ascii that you have to catch from the debug ccsip messages.

Hope this help.. for me.. saved the business.



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