Building CME with SIP vs. SCCP Phones

We currently have 4 CME Systems with SCCP phones. Recently, it was discuss by our upper management to start using SIP phones. Right now the system I have configured are using a PRI for dialtone with POTS lines for backup. What needs to different on the config to use SIP? Do I replace the 'telephony-service' and current 'dial-peers' or add to them?
This is all new to me and I still have not been able to attend any formal training.
Would there ever be a reason to use SIP and SCCP phones on the same system and would that even work?
Any help is appreciated!

Aha.... that's a different question.
The choice of your trunking protocol (PRI versus SIP versus something else like FXO/analogue) is largely independent of the protocol the handsets use. You can use SCCP for your handsets and then use SIP trunking into a provider for your external calls. The CME Router will handle the signalling translation between SCCP and SIP for you.
Like everyone else here, I would always use SCCP for the handsets. Whilst Cisco are improving SIP handset features all the time between releases you still can't do everything you that you can do with SCCP. You would also need version 15 to support some of the more useful SIP handset stuff including Extension Mobility, and my own personal view of the code quality on the version 15 trains would use such language that it can't be mentioned here.
SCCP to the handset and trunking over SIP is perfectly valid and I've used it many times with great success.
Hope this helps. Barry

Similar Messages

'voice-class codec' for SCCP phones (CME)?

Hi, with SIP phones it's possible to apply a codec voice class.
Let's say I have the following voice class:
voice class codec 1
codec preference 1 g722-64
codec preference 2 g711alaw
I can apply it for SIP phones, e.g. for pool 9:
voice register pool 9
voice-class codec 1
With SCCP phones, I can only set one codec with the 'codec' command under ephone.
My goal is to use 'codec transparent' in the dial peer and to let the phone itself negotiate the codec. How can I do this with SCCP phones?
For example, if I use 'codec transparent' in the dial-peer and someone (who doesn't support g722) calls me, then the SIP phone negotiates g711alaw with the other side and no transcoding is needed. This is what I also want for my SCCP phones. Am I missing a command?
I'm using CME 8.6

The syntax for SCCP phones is the same. Just apply the class to the VoIP dial peers.
dial-peer voice 100 voip
tone ringback alert-no-PI
description For InBound VoIP
modem passthrough nse codec g711ulaw
voice-class codec 1 <<<<<
voice-class h323 1
incoming called-number .
fax rate disable
no vad
Please rate helpful answers!

How do i monitor SIP Phone from a SCCP Phone? UC500

Hello Community.
This is my SIP Phone config:
voice register dn 2
number 214
name xxx
no-reg
label xxx
voice register pool 2
id mac 44E4.D944.9B11
session-transport tcp
type 9971
number 1 dn 2
number 2 dn 3
dtmf-relay rtp-nte
username xxx password xxx
codec g711ulaw
And this is the SCCP Phone:
ephone 12
device-security-mode none
mac-address 0011.2014.8248
ephone-template 16
username xxx password xxx
type 7960 addon 1 7914
button 1:12 2:51 8m11 9m10
button 10m15 11m13 12m16 13m17
button 14m18
Hope you can help, thanks Patrick

Yes you can.
Sample config :
sip-ua
          presence enable
     presence
          presence call-list
          max-subscription 64
          watcher all
          allow subscribe
     ephone-dn 12
          number 2010
          allow watch
          blf-speed-dial 1 214 label "Teset-User"
     voice register dn 2
          number 214
          allow watch
CME admin guide:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmepres.html#wp1011193
Please rate replies and mark question as "answered" if applicable.

CME/CUE SIP Phones DTMF-Relay

Hi all,
Just looking for some clarification on this one. I'm seeing some conflicting advice about setting the DTMF-Relay on SIP Phones registered to CME with a CUE Module. I've read some documentation indicating that rtp-nte RFC2833 is the only dtmf-relay supported for SIP Phones registered to CME, however I've also read some documents indicating that sip-notify must be configured as the dtmf-relay on SIP phones when they are communicating to a CUE module. I'm assuming I'm going to need to configure an MTP on the CME, but just wondering what the official DTMF config should be under the voice register pool for SIP phones.
Thanks!

Hi logan
When doing lab with cme 7.0 and sip phones .sip phones are not recognizing the "sip-notify" dtmf-relay method .It can only recognize "rtp-nte" method and it does not matter weather you are using sip-notify or rtp-nte for a dial-peer pointing to cme .
i configured on cue
ccn subsystem sip
dtmf-relay sip-notify
end
on cme i configured a dial-peer pointing to cue
dial-peer v 3888 voip
destination-pattern 3888
session target ipv4:177.3.11.10
codec g711ulaw
no vad
session protocol sipv2
dtmf-relay sip-notiy
on my sip phones
voice register pool 1
dtmf-relay sip-notify ------> now in this case cue wont recognize dtmf tones
when i change this dtmf-relay method to rtp-nte it recognizes dtmf tones to when recording a message

Cisco 877 router - Cisco IP phone won't register with SIP provider

Hi all,
I'm having a problem with a Cisco SPA504G phone not registering with the SIP carrier over the Internet. We've recently rolled out a Cisco 877 router onto a new NBN business connection and can't get the pre-configured IP phone to register.
When we tested the phone with the NBN-provided Netgear router, it worked fine, as it did with the previous Cisco 1841 router we were using on a different link.
The way it's setup is using VLANs to define the internal subnets, which are then assigned to the physical interfaces (since the 887 doesn't allow IP assignments to the interfaces directly).
VLAN 100 is the internal network and has a SBS2011 server – assigned to F0 – IP range is 192.168.1.0
VLAN 200 is the guest network and has Internet access only – assigned to F1 – IP range is 10.1.1.0
VLAN 500 is the WAN network and connects to the NBN upstream box – assigned to F3 – external IP address assigned by DHCP
I've been playing around with access lists, nat rules, basically everything in my limited Cisco knowledge to try and figure this out, but to no avail. I have even configured what I believe is unrestricted access to IP, UDP and TCP outbound and inbound to all VLANs and still can't get it to register.
Tried isolating the issue by creating a new VLAN and assigning it to the spare interface and basically allowing everything in and out, but still no luck.
The problem has to be something on the router – probably some small line of config I haven’t removed or added.
I am going to pull my hair out soon, so would really appreciate some assistance from the Cisco gurus out there.
My client has just purchased about 10 of these handsets from their provider so I need to fix this ASAP. The guy who provided them wasn't very helpful, and basically said I'm on my own once we tested using the NBN-provided Netgear router.
Happy to post my config as well.
Please help!!!!

Current configuration : 4912 bytes
version 15.1
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router1
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 10
crypto pki token default removal timeout 0
no ip source-route
ip dhcp excluded-address 10.1.1.1
ip dhcp pool GUEST
network 10.1.1.0 255.255.255.0
dns-server 10.1.1.1 203.50.2.71 139.130.4.4
default-router 10.1.1.1
ip cef
no ip domain lookup
ip domain name network.local
ip name-server 192.168.1.123
ip name-server 203.23.53.12
ip name-server 197.12.32.86
ip name-server 8.8.8.8
no ipv6 cef
license udi pid CISCO887VA-K9 sn FGL171220XY
username admin privilege 15 secret 5 $1$aNsm$N1BCQYkoi8gnURyvloYEX/
controller VDSL 0
interface Ethernet0
no ip address
shutdown
interface ATM0
no ip address
no atm ilmi-keepalive
bridge-group 10
pvc 8/35
interface FastEthernet0
description NAC - Internal network
switchport access vlan 100
no ip address
interface FastEthernet1
description NAC - Guest network
switchport access vlan 200
no ip address
interface FastEthernet2
no ip address
shutdown
interface FastEthernet3
description **** WAN Port ****
switchport access vlan 500
no ip address
interface Vlan1
no ip address
bridge-group 10
hold-queue 100 out
interface Vlan100
description NAC - Internal Vlan
ip address 192.168.1.1 255.255.255.0
ip access-group IN-100 in
ip access-group OUT-100 out
ip nat inside
ip virtual-reassembly in
interface Vlan200
description NAC - Guest Vlan
ip address 10.1.1.1 255.255.255.0
ip access-group IN-200 in
ip access-group OUT-200 out
ip nat inside
ip virtual-reassembly in
interface Vlan500
description **** WAN Vlan ****
ip address dhcp
ip nat outside
no ip virtual-reassembly in
no ip forward-protocol nd
ip http server
ip http access-class 23
ip http secure-server
ip dns server
ip nat inside source list NAT-100 interface Vlan500 overload
ip nat inside source list NAT-200 interface Vlan500 overload
ip nat inside source static tcp 192.168.1.123 25 interface Vlan500 25
ip nat inside source static tcp 192.168.1.123 443 interface Vlan500 443
ip nat inside source static tcp 192.168.1.123 3389 interface Vlan500 3399
ip nat inside source static tcp 192.168.1.123 80 interface Vlan500 80
ip nat inside source static tcp 192.168.1.123 4125 interface Vlan500 4125
ip nat inside source static tcp 192.168.1.124 3389 interface Vlan500 3390
ip nat inside source static tcp 192.168.1.123 987 interface Vlan500 987
ip nat inside source static tcp 192.168.1.123 1723 interface Vlan500 1723
ip route 0.0.0.0 0.0.0.0 55.234.52.43
ip access-list extended IN-100
permit udp any any range bootps bootpc
deny ip 10.1.1.0 0.0.0.255 any
permit ip 192.168.1.0 0.0.0.255 any
ip access-list extended IN-200
permit udp any any range bootps bootpc
permit ip 10.1.1.0 0.0.0.255 any
ip access-list extended NAT-100
deny ip 192.168.0.0 0.0.255.255 192.168.0.0 0.0.255.255
permit ip 192.168.1.0 0.0.0.255 any
ip access-list extended NAT-200
deny ip 10.1.0.0 0.0.255.255 10.1.0.0 0.0.255.255
permit ip 10.1.1.0 0.0.0.255 any
ip access-list extended OUT-100
permit udp any range bootps bootpc any
deny ip 10.1.1.0 0.0.0.255 any
permit ip any 192.168.1.0 0.0.0.255
ip access-list extended OUT-200
permit udp any range bootps bootpc any
deny ip 10.1.1.0 0.0.0.255 192.168.1.0 0.0.0.255
permit ip any 10.1.1.0 0.0.0.255
access-list 23 permit 59.23.164.52
access-list 23 permit 192.168.1.0 0.0.0.255
access-list 23 permit 10.1.1.0 0.0.0.255
access-list 23 permit 120.146.0.0 0.0.255.255
access-list 23 permit 149.185.12.0 0.0.0.255
access-list 23 permit 110.44.28.0 0.0.0.255
access-list 23 permit 110.44.26.0 0.0.0.255
access-list 23 permit 103.25.212.0 0.0.0.255
access-list 23 permit any
bridge 10 protocol ieee
banner motd ^C
* Authorized personnel only! *
^C
line con 0
login local
no modem enable
line aux 0
line vty 0 4
password password01
login local
transport input all
end

CME with linksys spa941

Hello, I have CME 4.0 with 2 IP Phones Linsys SPA941.The IP Phone register and functionally with the codec g711ulaw, but when use codec g729r8, the calls are tone busy. The codec is configuration on router and IP Phone. The configurations are attached.
Thank you.

Hi,
can they call each other in g729? can you call an SCCP phone if the latter is set for g.729 ?
Can you collect and send "term mon" and "debug ccsip message" when trying ?

Problems with a new IP phone 7906 after factory reset

Hi Guys
I am currently having problems with my new IP phone 7906 after doing factory reset and here is a video of it:
http://www.youtube.com/watch?v=vvCLtwIAwVw
I used this command I found on the net:
Hold down
while plugging in the phone and keep holding it until the message waiting indicator (red light) starts blinking. Then
release # and type 123456789*0# in sequence.
I was trying to switch my phone to sip and it also dont want to take tftp setup either.

Tim,
Found this on the net
http://www.cisco.com/en/US/ts/fn/620/fn62949.html
Might be woth giving it a go
6.Pull power on the phone (even if power is PoE).
7.Hold down the # key on the phone.
8.Continue holding down the # key and re-apply power.
9.While still holding the # key wait for the Message Waiting Indicator (MWI) light on the handset to start flashing.
10.Once the MWI light is flashing, release the # key and enter the following sequence exactly on the keypad:
3491672850*#
Once this sequence has been entered on the IP Phone, if all the network criteria above have been met, it should begin its recovery process. This process can take up to 15 minutes to finish. The phone may appear to be doing nothing during this time. However, if the phone does not recover after 20 minutes then it is possible that the recovery is stuck. In this case, re-examine your network and verify that steps 1-4 are in place, then re-issue the factory reset sequence.
* Note: The factory reset sequence is a way for a phone to clear flash and still upload to a valid firmware image. This is facilitated by the termxx.default.loads file, but requires that the image files listed in the termxx.default.loads file are available in TFTP for the phone to download. Open the termxx.default.loads file in any text editor. This loads file is essentially just a packing list showing all the OS and application files the phone needs to function. The files include a cnu, cvm, dsp, app and jar files. Please make sure that these files as listed in the termxx.default.loads file are in TFTP. ("xx" will be either "06" for the CP-7906G model, or "11" for the CP-7911G model.)
Additional Diagnostic Steps:
- Try a hard-coded IP address as a test to see if this resolves the upgrade failures. If it does, and the number of failing IP Phones is relatively few, this procedure may be the most expedient. After the IP Phone upgrades successfully, reconfigure the IP Phone to use DHCP.
- Try putting the phone on a hub or a different switch and see if this helps change the startup timing enough so that the upgrade completes successfully.
Regards
Alex

10.6.2: Bad image quality with SIPS

I used to convert my PDF documents with SIPS like this:
/usr/bin/sips --setProperty format jpeg --setProperty formatOptions high -z 400 200 sourcefilename.pdf --out targetfilename.jpg
This results in a very bad image quality after updating to 10.6.2 (the text contained in the source pdf file is nearly unreadable in the resulting image, especially when using the -z attribute for downsizing).
I could reconstruct this behaviour with several machines today: when using 10.6.1 the quality is fine, but after the update to 10.6.2 the resulting image quality is unacceptable.
Any ideas? Or could this be a bug? The man page for sips does not contain any information about new parameters and the image quality for other target formats (tif, png, ...) seems to be ok.

That picture looks like it was taken in a dimly lit room.
You could try using night mode but you will need a very steady hand.
Most basic phone cameras just cannot produce good pictures indoors when not in brightly lit areas. LED flashes just cannot do a good enough job when compared to real cameras.
Megapixels don't equal quality, it's the lens and flash that make the biggest difference.

Sccp phones unregistred

Hi all
I am upload 7937 firmware in telephony-service and recreate cnf files .
After that when restart the SCCP phones they can't registred in CME but receive DHCP IP addresses and CME IP .
I execute no load 7937 firmware and perform no cre cnf , cre cnf in telephony-service but its not helps me. Phones who not restarted working fine
sh telephony-service
CONFIG (Version=10.5)
=====================
Version 10.5
Max phoneload sccp version 17
Max dspfarm sccp version 18
Cisco Unified Communications Manager Express
For on-line documentation please see:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html
protocol mode default
ip source-address IP port 2000
ip qos dscp:
ef (the MS 6 bits, 46, in ToS, 0xB8) for media
cs3 (the MS 6 bits, 24, in ToS, 0x60) for signal
af41 (the MS 6 bits, 34, in ToS, 0x88) for video
default (the MS 6 bits, 0, in ToS, 0x0) for serviceservice directed-pickup
load 7916-24 cmterm-7916.1-0-4-2.cop.sgn
load 7965 SCCP45.9-4-2-1S
load 6921 SCCP69xx.9-2-1-0
max-ephones 35
max-dn 50
max-conferences 4 gain 6
dspfarm units 1
dspfarm transcode sessions 0
dspfarm 1 confprof1
conference hardware
privacy
no privacy-on-hold
hunt-group report delay 1 hours
hunt-group logout DND
max-redirect 10
cnf-file location: flash:
cnf-file option: PER-PHONE
network-locale[0] RU (This is the default network locale for this box)
network-locale[1] RU
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] RU (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
srst mode auto-provision is OFF
srst ephone template is 0
srst dn template is 0
srst dn line-mode single
moh music-on-hold.au
time-format 24
date-format dd-mm-yy
timezone 32 Russian Standard/Daylight Time
call-forward pattern .T
transfer-pattern .T
keepalive 180 auxiliary 30
timeout interdigit 10
timeout busy 10
timeout ringing 180
timeout transfer-recall 0
timeout ringin-callerid 8
timeout night-service-bell 12
caller-id name-only: enable
system message
web admin system name
web admin customer name Customer
edit DN through Web: enabled.
edit TIME through web: enabled.
background save interval 10 minutes
Log (table parameters):
max-size: 150
retain-timer: 15
create cnf-files version-stamp Jan 01 2002 00:00:00
transfer-system full-consult
transfer-digit-collect new-call
local directory service: enabled.
Extension-assigner tag-type ephone-tag.
Call List BLF is enabled
shutdown

I perform that
Router#configuration terminal
Router(config)#no logging console
Router(config)#no logging monitor
Router(config)#service timestamps debug datetime msec
Router(config)#logging buffered 40960 debugging
Router(config)#service sequence
Router(config)#no logging rate-limit
Router(config)#exit
debug tftp events
debug ephone detail
debug ephone register
In telephony-service i recreate cnf files and have this output :
Creating CNF files
IP address required is
TCP port required is 2000
read -1 bytes from flash:/its/SEPDEFAULT.cnf file
A0
0 item(s) of type 0
Unrecognized type 0 or format
Creating new SEPDEFAULT.cnf file size 58
58 bytes written OK..
ephone add http binding flash:/its/russia_lddefault.cfg failed
ephone add http binding flash:/its/russia_gkdefault.cfg failed
ephone add http binding flash:/its/russia_ffdefault.cfg failed
ephone add http binding flash:/its/united_states_lddefault.cfg failed
ephone add http binding flash:/its/united_states_gkdefault.cfg failed
ephone add http binding flash:/its/united_states_ffdefault.cfg failed
Just few seconds and i paste log
for DN 3 chan 1 to state CALL_END
009627: Mar 5 14:48:03.437: ephone-3[2/7]:UpdateCallState DN 3 chan 1 state 10 calleddn -1 chan 1
009628: Mar 5 14:48:03.437: ephone-3[2/7]:Binding ephone-3 to DN 3 chan 1 s2s:0
009629: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:Set FAC enabled (0) and dial mode (4)
009630: Mar 5 14:48:03.437: DN 3 chan 1 End Voice_Mode
009631: Mar 5 14:48:03.437: SkinnyGetCallState for DN 3 chan 1 IDLE
009632: Mar 5 14:48:03.437: called DN -1 chan 1, calling DN -1 chan 1 phone 3 incoming s2s:0
009633: Mar 5 14:48:03.437: ephone-3[2/7]:SkinnyStopMedia: normal line=1 dn=3 ch=1
009634: Mar 5 14:48:03.437: ephone-3[2/7]:SkinnyStopMedia: CloseReceive sent: normal confID=6 ref=1120
009635: Mar 5 14:48:03.437: SkinnyGetCallState for DN 3 chan 1 IDLE
009636: Mar 5 14:48:03.437: called DN -1 chan 1, calling DN -1 chan 1 phone 3 incoming s2s:0
009637: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:SkinnyStopMedia: Multimedia not active
009638: Mar 5 14:48:03.437: ephone-3[2/7]:SkinnyStopMedia: StopMedia sent: normal confID=6 ref=1120
009639: Mar 5 14:48:03.437: SkinnyGetCallState for DN 3 chan 1 IDLE
009640: Mar 5 14:48:03.437: called DN -1 chan 1, calling DN -1 chan 1 phone 3 incoming s2s:0
009641: Mar 5 14:48:03.437: ephone-3[2/7]:SpeakerPhoneOnHook
009642: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:Clean up activeline 1
009643: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:UpdateCallState unbind phone from DN 3
009644: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:ClearCallPrompt line 0 ref 0
009645: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:SkinnyCheckPendingCallBackPhone scan 2 lines
009646: Mar 5 14:48:03.437: ephone-3[2/7]:SelectPhoneSoftKeys set 0 mask FFBF for line 0 ref 0
009647: Mar 5 14:48:03.437: SkinnySetCallInfoName calling dn -1 chan 1 dn 3 chan 1,calling [] called []
009648: Mar 5 14:48:03.437: SetCallInfo DN 3 chan 1 is not skinny-to-skinny
009649: Mar 5 14:48:03.437: SkinnyStopDnRecallTimer: dn 3 chan 1
009650: Mar 5 14:48:03.437: Skinny Call State change for DN 3 chan 1 CALL_END from CONNECTED
009651: Mar 5 14:48:03.437: ephone-(3) DN 3 chan 1 calledDn -1 chan 1 callingDn -1 chan 1 :: port=0 incoming
009652: Mar 5 14:48:03.437: SkinnyUpdateCstate DN 3 chan 1 cstate 2
009653: Mar 5 14:48:03.437: SkinnyGetCallState for DN 3 chan 1 IDLE
009654: Mar 5 14:48:03.437: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009655: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:SkinnyUpdateCstate first phone for DN 3 chan 1 ref 1120
009656: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:UpdateCState found DN 3 on line 1
009657: Mar 5 14:48:03.437: ephone-3[2/7]:SkinnyUpdateCState process cstate 2 for inactive DN 3 chan 1 line 1 (activeLine=0 whisperLine=0)
009658: Mar 5 14:48:03.437: ephone-3[2/7]:SkinnyUpdateCstate inactive (overlay) line 1 for DN 3 ref 1120 combo=0
009659: Mar 5 14:48:03.437: DN 3 chan 1 ephone-3 state set to 2
009660: Mar 5 14:48:03.437: SkinnyGetCallState for DN 3 chan 1 IDLE
009661: Mar 5 14:48:03.437: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009662: Mar 5 14:48:03.437: ephone-3[7]:SetCallState line 1 DN 3(3) chan 1 ref 1120 TsOnHook
009663: Mar 5 14:48:03.441: ephone-3[2/7]:SkinnyTrackActiveCall for line 1 ref 1120 state 2 (slot 0)
009664: Mar 5 14:48:03.441: ephone-3[2/7]:SelectPhoneSoftKeys set 0 mask FFBF for line 1 ref 1120
009665: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009666: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009667: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009668: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009669: Mar 5 14:48:03.441: ephone-3[2/7]:Clean Up Speakerphone state
009670: Mar 5 14:48:03.441: ephone-3[2/7]:SpeakerPhoneOnHook
009671: Mar 5 14:48:03.441: ephone-3[2/7]:Speaker is not on, SpeakerPhoneOnHook suppressed
009672: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:SkinnyGetToneRef toneRef 0x0 callRef 0x460
009673: Mar 5 14:48:03.441: ephone-3[2/7]:SkinnyPhoneToneDirect: StopTone sent: normal line=1 ref=1120 tone=0x0
009674: Mar 5 14:48:03.441: Skinny StopTone sent on ephone socket [7]
009675: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009676: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009677: Mar 5 14:48:03.441: ephone-3[2/7]:SelectPhoneSoftKeys set 0 mask FFBF for line 0 ref 0
009678: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009679: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009680: Mar 5 14:48:03.441: ephone-3[7]:SetLineLamp 1 to OFF
009681: Mar 5 14:48:03.441: UnBinding ephone-3 from DN 3 chan 1
009682: Mar 5 14:48:03.441: ephone-3[2/7]:---SkinnySyncPhoneDnOverlays is onhook
009683: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009684: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009685: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:SkinnyArmPhoneCallbacks scan 2 lines
009686: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:ClearCallPrompt line 0 ref 0
009687: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:SkinnyCheckPendingCallBackPhone scan 2 lines
009688: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009689: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009690: Mar 5 14:48:03.441: SkinnyReportDnState for overlay DN 3 chan 1 on ephone-1
009691: Mar 5 14:48:03.441: SkinnyReportDnState DN 3 chan 1 ONHOOK
009692: Mar 5 14:48:03.441: ephone-3[2/7]:SkinnyConfirmOnHookAck: dn 3 chan 1 dn_index 3 phone=2, pickupOnHook=0
009693: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:ClearCallPrompt line 0 ref 0
009694: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:SkinnyCheckPendingCallBackPhone scan 2 lines
009695: Mar 5 14:48:03.445: dn_tone_control DN=3 chan 1 tonetype=0:DtSilence onoff=0 pid=418
009696: Mar 5 14:48:03.445: SkinnyGetCallState for DN 3 chan 1 IDLE
009697: Mar 5 14:48:03.445: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009698: Mar 5 14:48:03.445: ephone-3[2/7]:Check toneOn state for last_phone
009699: Mar 5 14:48:03.445: SkinnyGetCallState for DN 3 chan 1 IDLE
009700: Mar 5 14:48:03.445: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009701: Mar 5 14:48:03.665: ephone-3[2/7][SEP20BBC01F9387]:Update Stats Total for DN 3 chan 1
009702: Mar 5 14:48:03.761: ephone-3[2/7][SEP20BBC01F9387]:MediaPathEventMessage Handset OFF
009703: Mar 5 14:48:03.761: ephone-3[2/7]:MediaPathEventMessage

CME With CUE

Hey All ,
i have integrated My CME with CUE , i have two phone with extension 1000,2000 and all phones have mailbox,,
i make basic IVR that will forward the call to 1000 if 2000 is busy , but the case is when the 2000 is busy the forwarded call go to the mailbox
Not to 1000 .
how i can make the call go to 1000 NOT to the voice mail ??
but at the same time if any one is call the 1000 OR 2000 direct and the extension is busy will go the mail box.
please help Me...
Best Regards,
Ahmad Kefaya

thanks for pointing

Grave problems with SIP in N95 after update to v20

Hi,
I'm quite glad overall of updating my N95 to v20, but I've noticed a quite big bug in the new version.
When using the Internet Telephony client I can make SIP calls with no problem BUT if I receive a SIP call from a client with enabled video the phone hangs completely (I have to remove battery, no workaround).
This didn't happen with v12.
This is very annoying, because since I have no control over the settings of the softphones used by the people that call me, having my phone registered to Internet Telephony service offers a great risk of leaving me aout of service.
Is there any solution to this?
Thanks
Jorge

I am experiencing a similar issue after updating my iPad2 to iOS5. After the initial download/sync many albums were missing artwork and several dozen songs were greyed out and unplayable on the iPad (and an iPhone G3S with iOS5). However, upon resyncing a second time, most restored to normal cover art and function. HOWEVER, it remain weird in that when looking at Album view on the Music app all the artwork is present on the iPad, but it appears blank on Artist view for many artists. EXCEPT if I enter the list of Songs for a given artist the cover art reappears. This is clearly a glitch in how different views within Music are storing and displaying the artwork, as it is clearly there but not being proprely displayed. I am thinking about resyncing again, but hope Apple addresses this soon.
Thanks for any advice anyone can offer.

Lync 2013 with SIP trunk with panasonic kx-tde200

Hi
My company has installed a panasonic ip-pbx kx-tde for multiline with 100 number range for telephone service.
Now my company is going to replace multiline by sip trunk . It will still work with Panasonic pbx box just need to reprogramme to be able to connected to the sip proxy which is managed by internet service provider.
For this scenario , would Lync 2013 voice work if I just add PSTN gateway which is the ip of panasonic pbx address to the frontend in topology ? Or I may need a mediation server as a must requirement to make lync voice work?
Thanks
WenFei

Media bypass allow a call to basically skip the mediation server once it's established and go directly from gateway (in this case the PBX) and the endpoint (the telephone handset or Lync client) More information here: http://technet.microsoft.com/en-us/library/gg398719.aspx
By having this (if your PBX supports it) you reduce the load on the mediation servers. Before you go too far down the road also make sure that your PBX supports SIP trunks that are SIP over TCP (as Lync doesn't work with SIP over UDP)
Sort of, the easiest way is to add the .com as an additional SIP domain in Topology builder, you will need to create DNS records for it (both internal and external) and you will need to reissue the certs with additional SANs to support the second domain.
YOu will also need to update all the users to use the new suffix of xxx.com. So it's not a small task.
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Cisco ATA for Use with Time Warner Digital Phone

Is it possible to do the following with any of the Cisco products? Basically I want to have the incoming line from Time Warner go to the PBX to answer calls and then have the PBX route the calls appropriately. I currently have a Linksys PAP2 v2, but I don't think that will work because I think both ports are FXS and I'll need an FXO and an FXS. I have attached a diagram.
Regards,
Scott

So the Linksys PAP2 I have in my diagram, or a Cisco SPA-3102 cannot route inbound calls to the PBX via FXO and then route the call to internal analog phones via the FXS as directed by the PBX given that the PAP2 or SPA-3102 would register to the proxy? I'm really looking to just permit the PBX to talk to the analog stations since I can't get a SIP trunk from the Time Warner modem and with my lack of knowledge on VoIP it seems like the 2901 is over kill. If the router is necessary, then how is it that the PAP2 can connect directly to the internet and link up with Vonage or other similar providers with an analog phone connected to it? The PAP2 is going to communicate with Vonage via SIP and the phone is analog, so there must be some form of encoding going on inside the PAP2, which is what the PVDM does inside the router. So I guess I'm really looking for someone to solidify my understanding of what is needed that the PAP2 or SPA-3102 can't provide. There are a lot of posts on the internet with folks using these devices, but I suspect they are still using a SIP provider as opposed to an analog handoff like Time Warner gives.

Problem with sip trunk between CCM and Huawei through Cisco ASA5520

Hello,
I have a next problem
During SIP conversation ASA is changing the ip address of CCM to corresponding name in ASA configuration inside the SIP packet:
To: <sip:443230282@Server_CCM1;user=phone>
ASA name configuration:
name x.y.z.h Server_CCM1
But it should be without any changes like that: To: <sip:[email protected];user=phone>. Because of that session cant be established. Remote SIP peer gives an error "Bad Request - 'Malformed/Missing URL"
When name was deleted in ASA "no name x.y.z.h Server_CCM1" we have no any problem with SIP initialization and call proccesing.
We are going to upgrade ASA from 8.2 to 8.3 and it seems that we will have the same problem because object will be created automaticly in new version (we are using a NAT) and we will not be able to delete an object like we did in version 8.2.
What configuration in ASA version 8.3 should be done to avoid this issue.
P.S Detailed debug from Huawei in attachment.
Thank you.

Hi.
depending on your config, you might be hitting CSCta16361, this is fixed in 8.2(4)
if you can confirm it's still happening in latest 8.2 release, then a TAC case needs to be opened so investigation is done and a new bug is opened.
if you've tested 8.2(4) already and it's still doing the same, then a TAC Service Request should be opened for more investigation and possibly opening a new defect.
Best regards,
Fadi.
does this answer your question? if yes please mark it resolved.

CME with Microsoft Lync/Mediation server Integration

Hello
please, we want to integrate our Cisco CME with microsoft Lyn/Mediation Server.
Please, what do i need to configure on the Cisco CME?
Thanks

I have config on cme side. But not Lync side(
sip-ua
no remote-party-id
voice service voip
allow-connections sip to sip
no supplementary-service sip handle-replaces
sip
bind control source-interface gi0/0 (this should be your sip interface, creating loopback and binding to SIP is best practice)
bind media source-interface gi0/0 (this should be your sip interface, creating loopback and binding to SIP is best practice)
voice class codec 20
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
dial-peer voice 11 voip (Incoming dial-peer from Lync)
b2bua
session protocol sipv2
session target ipv4:LyncIP
session transport tcp
incoming called-number .%
voice-class codec 20
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 21 voip (outgoing dial-peer to Lync)
destination-pattern 1... (this is Lync patterns dial-peer)
notify redirect ip2pots
session protocol sipv2
session target ipv4:LyncIP
session transport tcp
voice-class codec 20
dtmf-relay rtp-nte
fax rate disable
fax protocol pass-through g711ulaw
no vad

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