Nortel CS1000E Release 7.5 using SIP to Cisco Unified Communications Manager 6.1.5

Actually exist diferents publications with other releases, but not with this options in common, somebody can help me???? Thank you!!!

I have a similar problem, only the tab with the Cisco Unified Communications Manager AXL servers option is disabled or appear disabled in CUC. any idea what might be going on, I checked that the service relates to AXL is on and running in CUCM and CUC.

Similar Messages

Ask the Cisco VIP: Troubleshooting SIP in Cisco Unified communications

Troubleshooting SIP in Cisco Unified communications deployments with Cisco VIP Ayodeji Okanlawon
This is a Q&A Ask the Expert Session continuation from the Live Webcast
Ask your questions on Session Initiation Protocol (SIP) and how it is redefining our UC world.The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
Featured Expert
Ayodeji Okanlawon, a Cisco Designated VIP, is the Lead Consultant Engineer for Global Solutions Design and Engineering at Verizon Business. In his past, he has worked at Intact IS, NCS Global, and Schlumberger Information Solutions. His experience includes development of design and deployment of large scale IP telephony projects on Cisco Call Manager platforms, Cisco Voice gateways, Cisco Jabber cloud and on premise solution. His expertise includes SIP solutions, CUBE design and Deployment, Troubleshooting: Voice gateways, CUCM, Unity connection, CUPS. Deji has been awarded the Cisco Designated VIP in 2013 and 2014. Deji holds a Bachelor of Science (BS), Electrical and Electronics Engineering, Second Class Upper from Obafemi Awolowo University.
According to Deji, “If you want to advance your career, if you’re serious about your skill sets, you’ve got to be in the forums.” (Read the Interview >>)
We look forward to your participation. This event is open to all, including partners.
* * Remember to use the rating system to let Deji know if you have received an adequate response. * *
Deji might not be able to answer each question due to the high volume expected during this event. This event runs January 13 through January 23, 2015. Visit this forum often to view responses to your questions and the questions of other community members.

Derrick,
RFC 3261defines ways to provide increased security for a SIP session.
The following describes areas in SIP that provides security for the protocol
1. Authenticating users.
We need to authenticate a user to ensure that the sender of the message is who he claims to be.
To achieve this SIP uses digest authentication between a UAC, proxy and a UAS. This provides the most basic level of authentication challenge between a client, proxy and a server.
2. Secure SIP signalling
The next area we can secure is SIP signalling itself. For this we use SSL/TLS. This is similar to using https in web browsers. With TLS before our any signalling is exchange X.509 certificates are used create a secure TLS channel. All our SIP messages are then transported within the secure channel.
NB: The digest authentication mentioned above for authenticating a user agent is just authentication. The messages are not protected from reading or modification hence it is recommended that these messages are carried inside a secure TLS channel for better security.
3. Privacy and Identification
Additional security features in SIP provides means where any user can choose to either reveal or conceal his identity.
4.Secure RTP
SIP also provides the ability to secure the media channel. It is not enough to secure signalling while anyone can listen to the media. RFC3830 discusses how the encryption should be done.
5. S/MIME
S/MIME encapsulation is used to protect sip headers making it impossible for any one in between the sender and receiver to modify the sip headers
Regards

Cisco Unified MeetingPlace Express Compatibility with Cisco Unified Communications Manager Release 8.X

I have a customer that would like to upgrade CUCM to either 7.1 or 8.0 and has Meeting Place Express.
In trying to establish compatability I found this link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/compat/ccmcompmatr.html#wp56211 which shows compatabilty up to CUCM 7.0.
If CUCM 7.0 is in fact the highest version supported with MPE 2.1(1.2), that would mean that they would loose their conferencing server.
Can anyone confirm if there will be MPE support for CUCM 7.1 or 8?
Thank you.

7.1 is still supported, however 8.X was never tested with MPE as it was already going to be replaced with MP.
Since the BU never tested it, it falls under not supported, MP8 is supported with CUCM 8.
But, since it uses a standard H323 trunk it should work fine.
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk

CME SIP issue - Cisco 7821 phone not registering

Hi
I am having issues with getting a Cisco 7821 phone to register.
Current deployment is with Cisco 6921 phones SCCP registration
SIP integration with CUE
SIP integration with Mitel system
c2951-universalk9-mz.SPA.154-3.M1.bin (CME 10.5)
In flash:
rootfs78xx.10-1-1SR1-4.sbn
kern78xx.10-1-1SR1-4.sbn
sboot78xx.10-1-1SR1-4.sbn
sip78xx.10-1-1SR1-4.loads
The 7821 phone gets IP address but fails to register. Please could somebody let me know why phone is not registering.
Configuration below (10.245.226.132 is CME address) .
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol pass-through g711ulaw
modem passthrough nse codec g711ulaw redundancy maximum-sessions 5
h323
sip
registrar server expires max 600 min 60
options-ping 90
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
voice register global
mode cme
source-address 10.245.226.132 port 5060
max-dn 30
max-pool 10
load 7821 sip78xx.10-1-1SR1-4
authenticate register
authenticate realm all
timezone 22
date-format D/M/Y
voicemail 590
tftp-path flash:
create profile sync 0061443538560005
network-locale GB
voice register dn 1
number 1010
name user1
label user1
mwi
voice register pool 1
busy-trigger-per-button 2
id mac F09E.636E.63F2
type 7821
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username 1010 password 123
codec g711ulaw
no vad
dial-peer voice 391 voip
description *** Auto Attendant ***
destination-pattern 399
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 392 voip
description *** Administration Via Telephone ***
destination-pattern 392
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 393 voip
description *** Extension Assigner ***
service ea out-bound
destination-pattern 393
session target ipv4:10.245.226.132
dial-peer voice 590 voip
description *** Voice Mail Pilot ***
destination-pattern 590
b2bua
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 1 pots
description ** Match all incoming POTS calls **
translation-profile incoming IncomingPSTNcalls
incoming called-number .
direct-inward-dial
dial-peer voice 899 voip
description Call to Mitel
translation-profile incoming Prefix9
translation-profile outgoing rem44
destination-pattern [23]..
session protocol sipv2
session target ipv4:192.168.114.2
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface GigabitEthernet0/0
description *** Connection to Mitel Phone System ***
ip address 192.168.114.5 255.255.255.248
duplex auto
speed auto
interface ISM0/0
description *** Connection to Cisco Unity Express ***
ip unnumbered GigabitEthernet0/1
service-module ip address 10.245.226.131 255.255.255.128
!Application: CUE Running on ISM
service-module ip default-gateway 10.245.226.132
interface GigabitEthernet0/1
description *** Connection to IP Phone LAN ***
ip address 10.245.226.132 255.255.255.128
duplex auto
speed auto
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 10.245.226.129
ip route 10.245.226.131 255.255.2
tftp-server flash:apps37sccp.1-4-4-0.bin
tftp-server flash:sip78xx.10-1-1SR1-4.loads
tftp-server flash:rootfs78xx.10-1-1SR1-4.sbn
tftp-server flash:sboot78xx.10-1-1SR1-4.sbn
sip-ua
mwi-server ipv4:10.245.226.131 expires 3600 port 5060 transport udp
registrar ipv4:10.245.226.132 expires 600
gatekeeper
shutdown
telephony-service
authentication credential cmeadmin c4p1ta2012
xml user xmladmin password xmladmin 15
extension-assigner tag-type provision-tag
max-ephones 104
max-dn 299
ip source-address 10.245.226.132 port 2000
auto assign 101 to 105
no service directed-pickup
timeouts interdigit 5
system message CFGS
url services http://10.245.226.131/voiceview/common/login.do
url authentication http://10.245.226.132/CCMCIP/authenticate.asp
cnf-file location flash:
cnf-file perphone
load 7931 SCCP31.9-2-1S
load 6921 SCCP69xx.9-2-1-0
time-zone 22
date-format dd-mm-yy
voicemail 590
max-conferences 8 gain -6
call-forward pattern .T
moh enable-g711 "music-on-hold.au"
web admin system name cmeadmin secret 5 $1$QmIK$46fDKVSudMxzI2bRp/Ef7/
time-webedit
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 298
number 598...
mwi on
ephone-dn 299
number 599...
mwi off

Page 7 of the following link recommends that you use option 150 with the Cisco 7800 series phones and use option 66 if you cannot use option 150
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7821_7841_7861/10_1/english/admin_guide/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0_chapter_01.pdf
Dynamic Host Configuration Protocol (DHCP)
DHCP dynamically allocates and assigns an IP address to network devices.
DHCP enables you to connect an IP phone into the network and have the phone become operational without your needing to manually assign an IP address or to configure additional network parameters.
DHCP is enabled by default. If disabled, you must manually configure the IP address, subnet mask, gateway, and a TFTP server on each phone locally.
Cisco recommends that you use DHCP custom option 150. With this method, you configure the TFTP server IP address as the option value. For additional supported DHCP configurations, go to the "Dynamic Host Configuration Protocol" chapter and the "Cisco TFTP" chapter in the Cisco Unified Communications Manager System Guide.
Note
If you cannot use option 150, you may try using DHCP option 66.

BCM interfacing to a Nortel CS1000E

Hi,
I was wondering if anyone has any experience with integrating a BCM with a Nortel CS1000E. The CS1000E is a VoIP PBX and is capable of SIP and H.323. I was wondering if I could use my CS1000E in lieu of a third party media gateway such as a Mediant 2000 SIP gateway for inbound and outbound calls to our voice network and the PSTN.
thanks in advance
Kevin
Edited by: KevinPrintz on Feb 18, 2011 1:02 PM

Hi David,
Have you tried starting the CDT with the loglevel=5 URL and looked at the client logs as well as connection server (bcmcs..) logs? Might help spot the problem.
See the SAP note 1753607 - BCM 7.0 How to change server and client loglevels if you need guidance with the log level settings.
BR,
-Lasse-

Issue with instant ringback when using sip trunk to SP

Hi all,
We use CUCM 8.0.2.
We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
c2900-universalk9-mz.SPA.150-1.M3.bin
Cisco CISCO2911/K9 (revision 1.0)
Technology Package License Information for Module:'c2900'
Technology Technology-package
                  Current       Type
ipbase        ipbasek9      Permanent
security      securityk9    Permanent
uc              uck9            Permanent
data           None            None
We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
We use 7945 and CIPC for our phones.
We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
Any ideas why this happens and how to stop it?
I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
voice service voip
address-hiding
mode border-element
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
header-passing error-passthru
early-offer forced
midcall-signaling passthru
interface GigabitEthernet0/0
ip address x.x.x.x 255.255.255.252
ip access-group acl.SIP-IN in
no ip redirects
no ip unreachables
ip verify unicast reverse-path
ip virtual-reassembly
duplex full
speed 100
no cdp enable
gateway
timer receive-rtp 1200
sip-ua
connection-reuse
gatekeeper
shutdown
dial-peer voice 1 voip
description *** INBOUND CALLS FROM CARRIER ***
translation-profile incoming SIPTRUNK-INCOMING
session protocol sipv2
incoming called-number #blah blah#
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 61 voip
description **** WA, SA AND NT NUMBERS ****
destination-pattern 0[8]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[8]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 81 voip
description **** MOBILE NUMBERS ****
destination-pattern 0[4]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[4]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 500 voip
description *** INBOUND SIP TRUNK TO CUCM PUB ***
translation-profile outgoing SIPTRUNK-CALLING-ADD-0
preference 1
destination-pattern 5[12]..
session protocol sipv2
session target ipv4:<OUR CUCM PUBLISHER IP>
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
Any help or a point in the right direction would be greatly appreciated.
Cheers,
Brett

I ended up resolving this issue as follows:
In CUCM, under Device > Device Settings > SIP Profile.
I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
Now, I get the expected delay before hearing ringback.
Solved!

Unable to call using sip communicator

i am using sip communicator.to run it i ve installed ant and jdk1.4 .also set the path.then i run the bat file of sip communicator and configure it also.bt i cnt established the call.thr is some errors.
again i am using my sip server.its domain name is bangla.net.in the sip-communicator.xml file i ve make some changes.its my xml file
<?xml version="1.0" encoding="UTF-8"?>
<configuration>
<log4j>
<rootLogger value="net.java.sip.communicator.common.Console.TraceLevel, RFLogger"/>
<appender>
<RFLogger value="org.apache.log4j.RollingFileAppender">
<layout value="org.apache.log4j.PatternLayout">
<ConversionPattern value="%r [%t] %p %c{2} %x - %m%n"/>
</layout>
<MaxBackupIndex value="1"/>
<File value="log/sip-communicator.app.log"/>
<MaxFileSize value="256KB"/>
</RFLogger>
</appender>
</log4j>
<net>
<java>
<sip>
<communicator>
<FIRST_LAUNCH value="false"/>
<ENABLE_SIMPLE value="false"/>
<media>

<PREFERRED_AUDIO_ENCODING value="0"/>
<PREFERRED_VIDEO_ENCODING value="26"/>
<MEDIA_SOURCE value=""/>
<MEDIA_BUFFER_LENGTH value="100"/>
<IP_ADDRESS value=""/>
<AUDIO_PORT value="22224"/>
<VIDEO_PORT value=""/>
</media>
<sip>
<PUBLIC_ADDRESS value="sip:[email protected]"/>
<TRANSPORT value=""/>
<REGISTRAR_ADDRESS value="192.168.110.33"/>
<USER_NAME value="20"/>
<STACK_PATH value="gov.nist"/>
<PREFERRED_LOCAL_PORT value=""/>
<DISPLAY_NAME value="pranti"/>
<REGISTRAR_TRANSPORT value="UDP"/>
<REGISTRATIONS_EXPIRATION value="3600"/>
<REGISTRAR_PORT value="5060"/>
<FAIL_CALLS_ON_DEST_USER_MISMATCH value="false"/>
<DEFAULT_DOMAIN_NAME value="bangla.net"/>
<DEFAULT_AUTHENTICATION_REALM value="bangla.net"/>
<WAIT_UNREGISTGRATION_FOR value="1100"/>
<SAME_USER_EVERYWHERE value="true"/>
<simple>
<CONTACT_LIST_FILE value="contact-list.xml"/>
<SUBSCRIPTION_EXP_TIME value="600"/>
<MIN_EXP_TIME value="120"/>
<LAST_SELECTED_OPEN_STATUS value="online"/>
</simple>
</sip>

<sipphone>
<IS_RUNNING_SIPPHONE value="false"/>
<MY_SIPPHONE_URL value="http://my.sipphone.com"/>
</sipphone>

<gui>
<AUTH_WIN_TITLE value="SIP Authentication!"/>
<AUTHENTICATION_PROMPT value="Please enter login name and password for the specified realm:"/>
<USER_NAME_LABEL value="User Name:"/>
<USER_NAME_EXAMPLE value="Example: 1-747-555-1212"/>
<PASSWORD_LABEL value="Password:"/>
<GUI_MODE value="PhoneUiMode"/>

<imp>
<CONTACT_LIST_X value=""/>
<CONTACT_LIST_Y value=""/>
<CONTACT_LIST_WIDTH value=""/>
<CONTACT_LIST_HEIGHT value=""/>
</imp>
</gui>
<common>
<PREFERRED_NETWORK_INTERFACE value="VIA Rhine II Fast Ethernet Adapter"/>
<PREFERRED_NETWORK_ADDRESS value="192.168.110.26"/>
</common>

<STUN_SERVER_ADDRESS value="stun01.sipphone.com"/>
<STUN_SERVER_PORT value="3478"/>
<VOICE_MAIL_ADDRESS value="17475551212"/>
</communicator>
</sip>
</java>
</net>
<gov>
<nist>
<javax>
<sip>
<SERVER_LOG value="log/sip-communicator.stack.log"/>
<TRACE_LEVEL value="16"/>
</sip>
</javax>
</nist>
</gov>
<javax>
<sip>
<IP_ADDRESS value="192.168.110.26"/>
<STACK_NAME value="sip-communicator"/>
<ROUTER_PATH value="net.java.sip.communicator.sip.SipCommRouter"/>
<OUTBOUND_PROXY value="bangla.net:5060/udp"/>
<RETRANSMISSON_FILTER value=""/>
<EXTENSION_METHODS value=""/>
<RETRANSMISSION_FILTER value="true"/>
</sip>
</javax>
<java>
<net>
<preferIPv4Stack system="true" value="true"/>
<preferIPv6Addresses system="true" value="false"/>
</net>
</java>
</configuration>
bt thr are still error
the errors are givenbelow.
net.java.sip.communicator.sip.CommunicationsException: Failed to create inviteTransaction.
This is most probably a network connection error.
     at net.java.sip.communicator.sip.CallProcessing.invite(CallProcessing.java:883)
     at net.java.sip.communicator.sip.SipManager.establishCall(SipManager.java:681)
     at net.java.sip.communicator.SipCommunicator.handleDialRequest(SipCommunicator.java:379)
     at net.java.sip.communicator.gui.GuiManager.dialButton_actionPerformed(GuiManager.java:342)
     at net.java.sip.communicator.gui.GuiManager$1.actionPerformed(GuiManager.java:612)
     at javax.swing.AbstractButton.fireActionPerformed(Unknown Source)
     at javax.swing.AbstractButton$ForwardActionEvents.actionPerformed(Unknown Source)
     at javax.swing.DefaultButtonModel.fireActionPerformed(Unknown Source)
     at javax.swing.DefaultButtonModel.setPressed(Unknown Source)
     at javax.swing.plaf.basic.BasicButtonListener.mouseReleased(Unknown Source)
     at java.awt.Component.processMouseEvent(Unknown Source)
     at java.awt.Component.processEvent(Unknown Source)
     at java.awt.Container.processEvent(Unknown Source)
     at java.awt.Component.dispatchEventImpl(Unknown Source)
     at java.awt.Container.dispatchEventImpl(Unknown Source)
     at java.awt.Component.dispatchEvent(Unknown Source)
     at java.awt.LightweightDispatcher.retargetMouseEvent(Unknown Source)
     at java.awt.LightweightDispatcher.processMouseEvent(Unknown Source)
     at java.awt.LightweightDispatcher.dispatchEvent(Unknown Source)
     at java.awt.Container.dispatchEventImpl(Unknown Source)
     at java.awt.Window.dispatchEventImpl(Unknown Source)
     at java.awt.Component.dispatchEvent(Unknown Source)
     at java.awt.EventQueue.dispatchEvent(Unknown Source)
     at java.awt.EventDispatchThread.pumpOneEvent(Unknown Source)
     at java.awt.EventDispatchThread.pumpEvents(Unknown Source)
     at java.awt.EventDispatchThread.pumpEvents(Unknown Source)
     at java.awt.EventDispatchThread.run(Unknown Source)
Caused by: javax.sip.TransactionUnavailableException: Could not resolve next hop or listening point unavailable!
     at gov.nist.javax.sip.SipProviderImpl.getNewClientTransaction(SipProviderImpl.java:351)
     at net.java.sip.communicator.sip.CallProcessing.invite(CallProcessing.java:876)
please tell me wht kind of error it is.why i cnt make the

Did you find out what caused the error??

Error release purchase order using transaction code me29n

i have warning error this purchasing document number cannot be released.
while i enter me29n t code.why?
then how to release purchase order using me29n t code?
can anyone answer this two questions.

hi
i have warning error this purchasing document number cannot be released. while i enter me29n t code.why?
is the release strategy is triggred for the po ?? check thisin P order in me23n ,in header level check release strategy tab ,
then how to release purchase order using me29n t code? can anyone answer this two questions.
just go to me29n >release strategy tab here just click on the release button in front of the release code

X5800 how to use SIP and VOIP

Hello, I have a X5800 music, model 5800d-1 and I can enter all the necessary SIP data. I get the message (settings, connectivity, admin settings, sip settings) registered which is fine. My provider is freephonie. Question: 1) Although on the freebox the SIP is activated, I don't receive incoming call on my nokia. 2) How do I tell my nokia to call on the internet ? Is it possible only using SIP tel numbers which I can enter in the contacts list? Help much appreciated. Pierre.

SIP and VoIP are not synonymous. VoIP can indeed use SIP (and often does), but other services, such as video sharing for example, also use SIP.
It so happens that the 5800 has no native support for VoIP, which is why you can't make SIP calls using it.
Your only solution for SIP is to use third-party services such as fring, for example. Once logged into your fring account you can subscribe to a SIP service. Fring's systems will register to the SIP account and relay the comminucation through your fring account. There's lag and the audio quality isn't as good as a direct SIP connection but at least it works.
The alternative is to use a phone that supports VoIP/SIP natively.
Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you!

Error when releasing the WS using the transaction WSCONFIG

Hello Colleagues,
Following error message is recieved when i try to release the webservice using the transaction WSCONFIG.
Error when creating Web service (Method call of IF_SRT_REGISTRY_ENTRY~CREATE_NEW failed; the class does not exist)
Can you please let me know if anyone of you have faced the similar error and have been successful in solving.
Thanks & Best regards,
Ajay

Hi Ajay,
Check if the thread below resolves your issue:
[Re: web service - How activate host ?;
Regards, Trevor

Nokia N95 Hangs while using SIP

Hi
I have Nokia N95 with v 20, I am facing hang problem every time when i use SIP/VOIP application to call or receive call after 15 to 20 minutes mostly phone hangs and need to remove the battery and on it again....
any body have any solution for this hang issue...? please advice
Regards
Saqib

Go into your control panel, System > Hardware > Devices, look for the USB (Universal Serial Bus Controllers)
[+]USB
Generic USB hub
Bleah
Bleah
Simply right mouse click on all of them select "uninstall" once they've all been removed, restart your machine, windows will re detect all your USB devices, and hopefully this time windows will install either a newer or older version, different version of drivers.
Then re-install..... there's not a lot else you can try.

How to develop VoIP client using SIP in J2ME?

Hi Everybody,
I want to develop a VoIP client in J2ME that connects to asterisk server of debian and can call to the registered user of asterisk server and can have a telephonic talk session easily.
Do anybody have idea regarding the development of the client or having tutorial that teaches the development of VoIP in J2ME or in any other way.?
PLZ help me to provide the solution.
Thanks in anticipation.
with regards,
KHAKHAR SAGAR

Hi
I am interested about developing VoIP application (using SIP) in J2ME platform. But I am stuck with the problem of MMAPI. Without using MMAPI J2ME has no access to mobile media devices, such as speaker or microphone, and without creating a player MMAPI can't play media data, such as sound or video. But its not possible to record voice and play voice data simultaneously using player in J2ME. So it seems almost impossible to implement VoIP application maintaining all its constraints and requirements, specially in case of delay and jitter.
I am looking for some solution, which will provide the ability to overcome this problem. I come out with two possible solutions, but not sure about their out come. If we can develop a native media application, we can have access to it by using KNI (K Native Interface). In that way we can take some risk to develop VoIP application for J2ME. My another solution is, we can handle the player using MMAPI to record and play voice data in mill second level, so that we can have a real time feeling, though I am not sure if its possible by using RTSP.
If any one have solution of this problem, please help us.
Reagards
Asif Mohammed Adnan

Released for external use RFMs

Hello,
Is there a comprehensive list of all the RFMs marked as "released for customer/external use" somewhere? If not, is there a way to find if a particular RFM is or not released for external use?
BTW, SAP Interface Repository was not a great help since it has information only on selected RFMs.
Thanks.

Hello,
Thank you for the answer, Christoph. There is definitely a field in the attributes of a FM which says "Released on" or "Not released". I hope you are talking about this field.
A couple of more things:
So, there are very few FMs which are actually released for external use - most of these being BAPIs. Is that right?
(ii) I can not use an FM which says "Not released" in my app, if I want certification of it - right?? It has to say "Released on: date of release".
Thanks.

WIP310-G2: how to connect to an specific VoIP provider using SIP ?

Dear all,
I just received a WIP310-G2 and I am wondering how -or whether- I can configure it to connect to an specific Internet based VoIP provider using SIP
Cant seem to find how to do it....
Thanks a lot,
Alvaro

You connect your WIP310 to your PC using an USB cable. Then lookup the IP-adresse on the phone and enter that into a browser.
In the UI select Admin Logon and you can enter all the configuration details for your VoIP provider. I would recommned that you ask your VoIP provider to help you with a guide.
I used a PAP2T configuration example that worked for my WIP310, which I got from my VoIP provider.

Can't establish a Voice gateway (cisco 2911) using SIP with CUCM 9.1

I have configured a Cisco 2911 as a Voice Gateway using SIP (the configuration is attached), but unfortunately can't establish a test call to a phone (CUPC 8.6 SCCP) using csim start. I have done logging the ccsip debug and ccapi debug and attached them. Could anyone help me to solve this problem?

I just did some research on my end and csim is not supported for SIP. The Invite will never be created and sent to the CUCM to initate the call. It disconnects in the router itself with normal cause.
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/sipSPIOutgoingCallSDP:
Could not create source SDP for Outgoing Call
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/sipSPICreateOutboundSDP:
Error in creating an SDP for the outbound call - Check for supported codecs
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/preprocessSetup:
Error during outbound SDP creation
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(16) for outgoing call
Please use an actual call to test your dial-peer and integration with call manager. csim will not work.
Hantale
Sree

Nortel CS1000E Release 7.5 using SIP to Cisco Unified Communications Manager 6.1.5

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