SIP Refer problem

Hi Guys,
I have problem at one of our customer side. When an operator received a call, and try to do a transfer (consult transfer) to a colleague, the colleague's phone ring one time and the call failed. The operator have to take back the call and give the full external number of the colleague to the caller.
After a reboot of the phone (unplugged/plugged) the transfer work fine !
I tried to find the cause in the CUCM traces but I need help.
Saw that Refer messages:
14:49:26.294 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 22.212.126.35 on port 50108 index 227
[21667403,NET]
REFER sip:[email protected]:50108 SIP/2.0
Via: SIP/2.0/TCP 22.214.10.18:5060;branch=z9hG4bKf63095b1b37b1
From: <sip:[email protected]>;tag=435147058
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 REFER
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=tcp>
User-Agent: Cisco-CUCM8.6
Require: norefersub
Expires: 0
Refer-To: cid:[email protected]
Content-Id: <[email protected]>
Content-Type: application/x-cisco-remotecc-request+xml
Referred-By: <sip:[email protected]>
Content-Length: 341
<x-cisco-remotecc-request>
<answercallreq>
    <dialogid>
      <callid>[email protected]</callid>
      <localtag>6511241~ada938a2-d3e6-4928-be65-07fc5a6e0f23-49485024</localtag>
      <remotetag>3cce73acbe00bd3fda57ded7-91adfcf9</remotetag>
    </dialogid>
</answercallreq>
</x-cisco-remotecc-request>
|2,200,21,1.3256060^22.214.10.2^SEP3CCE73ACBE00
After the OK:
14:49:26.317 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 22.212.126.35 on port 50108 index 227 with 442 bytes:
[21667404,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 22.214.10.18:5060;branch=z9hG4bKf63095b1b37b1
From: <sip:[email protected]>;tag=435147058
To: <sip:[email protected]>;tag=3cce73acbe00bd40ed051b81-4b2d068b
Call-ID: [email protected]
Date: Thu, 07 Feb 2013 13:49:17 GMT
CSeq: 101 REFER
Server: Cisco-CP7962G/9.3.1
Contact: <sip:[email protected]:50108;transport=TCP>
Content-Length: 0
|3,100,63,1.7871803^22.212.126.35^*
And thats the last messages with call ID starting by: 2e587c80
So could you say me how I can find the Transfer "destination" and why the call failled ?
Thanks,
Hervé Jacquemin

Hi hjacquemin,
Have you solved this problem? I found the same thing with my customer. We are using
6921 with SIP Firmware. It is very weird that my IP Phone was hanged and must reset by unplug and plug.
Winai.

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Partitions:
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- CiscoEndpoints
Calling Search Space: "ThirdParty_Cisco" contain both of the above partitions.
Route Pattern 531XX and 80965 are assigned to Route Partition "ThirdPartyPBX"
Cisco UCM Main site phones are in CSS "ThirdParty_Cisco" and DN is in Route Partition "CiscoEndpoints". DN is in CSS "ThirdParty_Cisco".
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Hello, I have a 2801 router that has been configured with CME and a working SIP connection to my local ISP.
Tested with calls via CME so I know for sure that the SIP config and dial plan is fine on this gateway.
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ps. I have attached a couple of screenshots of my config.

Hello, thanks for helping.
I activated "debug voice ccapi inout" as well as "debug ccsip all" on the gateway but nothing showed up.
Therefore I deduce the call is not even making it to across the SIP trunk into the gateway router ?
As I am a newbie trying this out for the first time, it is guranteed to be something really simple.
I have included my running config from the gateway router below..
One addition I made was to add an incoming dial peer. That is "dial peer 5, description CUCM SIP trunk".
I set it up with a destination patter 2... to match my phone config on CUCM which have numbering in the 2000 range.
Sorry, I got RTMT up and running but could not get any meaningful results from it. I need to learn up on that.
I did however run a 'dialed number analysis' from CUCM direct and have attached the result. It seems the dialled number "99" is matching the route pattern OK.
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Thanks if you guys can offer any more help.
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot system flash:c2801-ipvoicek9-mz.151-2.T0a.bin
boot-end-marker
no aaa new-model
clock timezone nzst 13 0
dot11 syslog
ip source-route
ip dhcp pool DATA_SCOPE
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   default-router 192.168.200.1
   dns-server 8.8.8.8
ip dhcp pool VOICE_SCOPE
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   default-router 192.168.100.1
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ip dhcp pool MGMT_SCOPE
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ip cef
ip name-server 4.2.2.2
no ipv6 cef
multilink bundle-name authenticated
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729r8
codec preference 3 g711ulaw
codec preference 4 ilbc
voice translation-rule 1
rule 1 /^9/ //
voice translation-profile Strip9ToGetOut
translate called 1
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-2995340181
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2995340181
revocation-check none
crypto pki certificate chain TP-self-signed-2995340181
certificate self-signed 01
3082023E 308201A7 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 32393935 33343031 3831301E 170D3733 30363034 31393534
32305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 39393533
34303138 3130819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100C34D C8ECBB53 E01373A3 2E286B78 2D23042B 1C8588B1 A7861899 BA1C6860
AE1D7868 2A59E3BC 54D0A457 8FFDE27F C09104E5 C7A429F3 74CD9DA8 4A980366
675CC27C CDB94838 821CC05F 2C0AC2BC D882C132 6CAA1FA6 6DA740E4 562428B1
12B741F1 A50C9246 4CC35EDA DEE1D038 3883BB35 A91ABF8B 483E4160 F5FA4B5A
9A570203 010001A3 66306430 0F060355 1D130101 FF040530 030101FF 30110603
551D1104 0A300882 06526F75 74657230 1F060355 1D230418 30168014 72119640
F3396E1F E4168086 D31D8619 0D8337FF 301D0603 551D0E04 16041472 119640F3
396E1FE4 168086D3 1D86190D 8337FF30 0D06092A 864886F7 0D010104 05000381
81003B5A 29DE3A1E C5AB6092 E8D90650 C80752FC 0AAC93FD C5DE3D69 071B08FA
D4013232 81CA07E7 15F90190 6A3AD6A0 1D05F0F2 13479568 888332A5 F81E2681
7DA44095 4D11CFB7 CA79579A 8D95DE54 7B00173C E2C50573 A310C8C9 1487FEFC
CE35B66E 9EF94CFA 8D6D6DCD ADC78132 2709F198 6DF2F0FA D80CC088 D0C4C7D1 080B
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duplex auto
speed auto
interface FastEthernet0/1
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interface FastEthernet0/1.2
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tftp-server flash:/phone/7940-7960/P00307020200.loads alias P00307020200.loads
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tftp-server flash:/phone/7940-7960/P00307020200.sbn alias P00307020200.sbn
control-plane
mgcp fax t38 ecm
dial-peer voice 1 voip
description local_7_Digit_Calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 9[2-9]......
session protocol sipv2
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voice-class codec 1
dial-peer voice 2 voip
description international_calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 900T
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 3 voip
description national_calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 90[34679].......
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 4 voip
translation-profile outgoing Strip9ToGetOut
destination-pattern 90[34679].......
dial-peer voice 5 voip
description CUCM SIP trunk
destination-pattern 2...
session protocol sipv2
session target ipv4:192.168.2.115
voice-class codec 1
sip-ua
authentication username xxxxxxxxxx password xxxxxxxx
060
telephony-service
max-ephones 10
max-dn 20
ip source-address 192.168.1.99 port 2000
load 7960-7940 P00307020200
max-conferences 4 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 1000
name Lydia Francis
ephone-dn 2 dual-line
number 1001
name Leah Francis
ephone-dn 3 dual-line
number 1002
n
ephone-dn 4 dual-line
number 1003
ephone 1
mac-address C80A.A970.01DE
type CIPC
button 2:2
ephone 2
mac-address 000C.3070.8705
button 1:1 2:15
ephone 3
mac-address 000C.8546.5954
button 1:3 2:15
line con 0
logging synchronous
line aux 0
line vty 0 4
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ntp server 195.43.74.123
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hai muhammad..
A required part of a clause or expression has been omitted. For example, a SELECT statement may have been entered without a list of
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SELECT TABLE.
Action:
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here we can use like that muhammad..
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bye now..
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Developer wrote in message <[email protected]>...
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</session>
<session>
<ejb-name>A</ejb-name>
<home>com.xxx.AHome</home>
<remote>com.xxx.A</remote>
<ejb-class>com.xxx.ABean</ejb-class>
<ejb-ref>
<ejb-ref-name>ejb/CBean</ejb-ref-name>
<ejb-ref-type>Entity</ejb-ref-type>
<home>com.xxx.CHome</home>
<remote>com.xxx.C</remote>
<ejb-link>C</ejb-link>
</ejb-ref>
</session>
</enterprise-beans>
<ejb-jar>
In this case, upon starting the JavaXAServer I get a
javax.naming.NameAlreadyBoundException thrown. The EJB spec
specifically states that I should be able to refer to C with the same
name from A and B. Is this a bug?
I can get around this by changing the name of the second reference to
something like "ejb/CBean2".
Additionally, I get a NameNotFoundException when looking up the
reference from within A or B as follows :-
Context ic = new InitialContext();
Object ob = ic.lookup("java:comp/env/ejb/CBean");
The only way I can work around this, is to construct the
InitialContext using the 'client' view, passing
com.beasys.jndi.WLEInitialContextFactory as the factory name, passing
the ISL host/port in the PROVIDER_URL and then using the JNDI names
for the lookup.
What's going wrong?
TIA

Nokia N91 SIP Configuration (problem)

Hi all,
As it ays in the subject, I own a nokia N91 handset and i'm having problems accessing SIP Settings.
Basically when I click on SIP Settings, and view SIP providers. I cannot delete, edit, or add any settings at all! When I do select (for example), options - delete - delete confirmation (yes) The SIP profile doesn't delete.
I have done a full factory reset on the phone (*#7370#), but still no luck

19-Apr-2008 06:11 PM
os2lover wrote:
Hi,
No the *#7370# doesn't affect your harddisk, only the phone memory.
Best regards,
Olivier Baum
ohh thanks for this info... Im just curious, how do we format the hard drive? Thanks
CeS
"The best index to a person's character is how he treats people who can't do him any good, and how he treats people who can't fight back"

Ref: problem after installing patch3 of 9ids

correction:
browser's status bar shows 'done' and nothing else occurs..

Hi,
Patch3 obviously conains another version of JInitiator and you are using IE as a Browser. You need to change the following lines in formsweb.cfg with the new information
jinit_classid=clsid:CAFECAFE-0013-0001-0009-ABCDEFABCDEF
jinit_exename=jinit.exe#Version=1,3,1,9
jinit_mimetype=application/x-jinit-applet;version=1.3.1.9
IE is very sensitive about the classid used with Jinitiator.
Check "Program Files"-->Oracle --> Jinitiator <version> --> doc --> jinit-version.ini for the calls id expected by teh new Jinitiator.
if you haven't installed the latest Jinitiator, just make sure, that the classid entry in formsweb.cfg file contains what the client Jinitiator version expects.
Hope this helps you solving the problem as here and then does for me.
Fran

Sip refer getting 404 & 503

Hi All,
We have a cisco 2911 running 15.0(1r)M6. We have configured sip trunks within CUCM 7.1(5) to the sip router single ip address of 192.168.10.51. When we use the AT&T trunk to make a sip call, everything works as expected (inbound and outbound calls).
We are trying to add a directory handler server (paralance) into the mix. It's ip address 10.13.115.140. The idea behind this system is to call it,tell it who you are looking for, and the call is transfered. So far the system accepts the call, confirmed who you want to be transfered too, and attempts the transfer.
During the transfer, the attached debug is getting us 404, and 503 responses. The caller hears dead air.
Ip Address of phone: 10.17.1.133. this is a 7941 registered to CUCM 7.1(5)
Ip Address of directory server. 10.13.115.140.
Ip address of router. 192.168.10.51
The calling number - 13304
The called number 12144 - routes to the sip trunk, then hits the dial peer.
The number being asked for 13157
dial-peer voice 296 voip
description CUCM Internal Dialing to CMS1
preference 1
destination-pattern 1....$
session protocol sipv2
session target ipv4:10.13.115.6
session transport udp
voice-class codec 299
dial-peer voice 297 voip
description CUCM Internal Dialing to CMS2
huntstop
preference 2
destination-pattern 1....$
session protocol sipv2
session target ipv4:10.13.120.6
session transport udp
voice-class codec 299
dial-peer voice 2912411 voip
huntstop
preference 1
destination-pattern 12411
session protocol sipv2
session target ipv4:10.13.115.140
session transport udp
voice-class codec 299
dtmf-relay rtp-nte
voice class codec 299
codec preference 1 g711ulaw

This could by one of three things:
1. Your softswitch is setup incorrectly.
2. "789096120445" is not a dialable number.
3. "1026" is not an acceptable caller-id. (this is less likely, I would expect a different error -- but still, is this caller-id allowed?)

CUCM 10.5.1 and Exchange 2010 Unified Messaging (UM) SIP Trunk Problem

This is more a comment if you're migrating from a lower version to 10.x. Hopefully Google will pickup this post so others don't spend too much time (I got lucky and found this in about 30 minutes).
There are many more SIP options than in the past. If you configured your integration as per the integration doc, all settings are relevant, however there are some new defaults that need to change.
SYMPTOM: Dialing another number or AA pilot and being redirected internally works, but the call drops on calls from external phones. Exchange logs an Event ID 1079 from UMCore and also informational 1084 and 1172 events. A capture yields a status 200 OK, then an ACK, two BYEs and a status 481 that the call leg does not exist. The call is then dropped.
RESOLUTION:
In the SIP Trunk, modifiy the following:
Device Information. . . Check Media Termination Point Required. I've had some that have needed this checked and some where I needed to leave unchecked. In this case going from 7.1 to 10.5 necessitated enabling.
Call Routing Information. . . SIP Privacy: Need to change from Default to None.
Any comments how the SIP Privacy might affect security and functionality would be appreciated.

Just to inform everyone. I rebuilt the edge server from scratch.
Now everything is working as expected.
I cannot work out how the edge was not passing on calls to exchange or not communicating correctly with exchange.
Anyway it is resolved now.

Sip connection Problem

hello everyone Please help..its urgent
i am a beginner in j2me..M trying to register with sip server but everytime it returns 0 response code and in wireshark its shows 401 one time after that 100 all the t ime continously..
how to solve this prblem..i have used oracle code as it is ..jsr 180 code but not able to get 200 ok response code ..
please help
Thanks
also mail me at [email protected]

Just to add:
I can telnet to 5060 to both ends of the SIP trunk, to Call Manager and to the CME

SIP Refer problem

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