CCME with Linksys SPA SIP phones

Dear all,
I'm currently working out with a deployment which's using the CME (not sure what version going to purchase at the moment)and using the Linksys SIP phone (SPA922). I would like to know whether the SIP phone's is able to register to CME or using the features that provided by CME. I did try to search from Cisco website but couldn't find any related information between two different hardware setup.
Thanks

Is there any documentations from Cisco which stated clealy that the CCME integration with different brand of SIP phone (at least Linksys) because i don't want to be happen that i buy a "bunch" of Linksys SIP phones and it cannot be integrate with CCME.
It should be better that to have a clear stated docs and can show to the management dept before purchasing.
Any ideas?
Thanks

Similar Messages

200 OK message before call is established with linksys SPA 3102

I recently bought a cisco linksys SPA 3102 gateway to help me forward incoming VOIP calls to the PSTN network via the PSTN line. I also installed syslog to catch the sip trace. When i placed a call, after the SIP Invite and Trying, I immediately get a 200 OK reply from the PSTN LINE, just as soon as the calls is forwarded to the PSTN network for dialing. This 200 OK reply triggers the billing from the SIP side mean while the call has not yet been established.
Is there a way to stop this per-matured 200 OK reply from happening?
I will be very grateful for your help or hints.
Cheers
Emmanuel

I recently bought a cisco linksys SPA 3102 gateway to help me forward incoming VOIP calls to the PSTN network via the PSTN line. I also installed syslog to catch the sip trace. When i placed a call, after the SIP Invite and Trying, I immediately get a 200 OK reply from the PSTN LINE, just as soon as the calls is forwarded to the PSTN network for dialing. This 200 OK reply triggers the billing from the SIP side mean while the call has not yet been established.
Is there a way to stop this per-matured 200 OK reply from happening?
I will be very grateful for your help or hints.
Cheers
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How to use TSP with CME9.1 + SIP Phones + EM

Hi,
can anyone explain me how to use the following setup:
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Kind regards
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Cisco SIP Phone 9971 won't register on CME 8.6 or 8.5 Please HELP

Please help me , I have problem with registering Cisco SIP phone 9971 with CME 8.6 on ISR 2901.
I configured CME for SIP clients, then I add configuration for 9971 phone and create profiles. Phone downloaded SEP...xml file from CME,after that phone look for g4-tones.xml and gd-sip.jar files, I added them to CME after that phone downloaded them and reboot. Now phone is stuck in some kind of loop and does not register on CME.
On phone log I can see repeting next few messeges.
12:01:58a No DNS Server IP
12:01:59a Updating Trust list
12:01:59a No Trust List instaled
12:01:59a SEP04C5AB03B0D.cnf.xml (TFTP) // at this time phone download SEP...xml file from CME
12:02:00a VPN Error: VPN is not Configured
on CME if issue DEBUG TFTP EVENTS i receive next few lines
*Aug 18 18:20:19.891: TFTP: Looking for CTLSEP04C5A4B03B0D.tlv
*Aug 18 18:20:19.987: TFTP: Looking for ITLSEP04C5A4B03B0D.tlv
*Aug 18 18:20:20.083: TFTP: Looking for ITLFile.tlv
*Aug 18 18:20:20.347: TFTP: Looking for SEP04C5A4B03B0D.cnf.xml
*Aug 18 18:20:20.351: TFTP: Opened flash:/SEP04C5A4B03B0D.cnf.xml, fd 14, size 4585 for process 141
*Aug 18 18:20:20.363: TFTP: Finished flash:/SEP04C5A4B03B0D.cnf.xml, time 00:00:00 for process 141
here you can see verison info of CME
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.1(4)M, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2011 by Cisco Systems, Inc.
Compiled Thu 24-Mar-11 15:31 by prod_rel_team
ROM: System Bootstrap, Version 15.0(1r)M9, RELEASE SOFTWARE (fc1)
ELTOSAN_ROUTER uptime is 1 hour, 50 minutes
System returned to ROM by reload at 16:29:20 UTC Thu Aug 18 2011
System image file is "flash:/c2900-universalk9-mz.SPA.151-4.M.bin"
Last reload type: Normal Reload
Last reload reason: Reload Command
Cisco CISCO2901/K9 (revision 1.0) with 471040K/53248K bytes of memory.
Processor board ID FGL1508252Y
3 Gigabit Ethernet interfaces
2 terminal lines
1 Virtual Private Network (VPN) Module
4 Voice FXO interfaces
4 Voice FXS interfaces
1 Internal Services Module (ISM) with Services Ready Engine (SRE)
   Survivable Remote Site Voicemail (SRSV) on Cisco Unity Express (CUE) 8.5.1 in slot/sub-slot 0/0
DRAM configuration is 64 bits wide with parity enabled.
255K bytes of non-volatile configuration memory.
254464K bytes of ATA System CompactFlash 0 (Read/Write)
License Info:
License UDI:
Device#   PID                   SN
*0        CISCO2901/K9          xxxxxxxxxxxxx
Technology Package License Information for Module:'c2900'
Technology    Technology-package          Technology-package
              Current       Type          Next reboot
ipbase        ipbasek9      Permanent     ipbasek9
security      securityk9    Permanent     securityk9
uc            uck9          Permanent     uck9
data          None          None          None
Configuration register is 0x2102
this is RUNNING CONFIGURATION
! Last configuration change at 16:10:12 UTC Thu Aug 18 2011
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname ELTOSAN_ROUTER
boot-start-marker
boot system flash:/c2900-universalk9-mz.SPA.151-4.M.bin
boot-end-marker
no aaa new-model
no ipv6 cef
ip source-route
no ip routing
no ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.5.1 192.168.5.10
ip dhcp excluded-address 192.168.5.200 192.168.5.255
ip dhcp pool phone
   network 192.168.5.0 255.255.255.0
   default-router 192.168.5.251
   option 150 ip 192.168.5.251
ip dhcp pool data
   relay source 192.168.2.0 255.255.255.0
   relay destination 192.168.2.201
multilink bundle-name authenticated
crypto pki token default removal timeout 0
voice-card 0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol pass-through g711alaw
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 192.168.5.251 port 5060
max-dn 6
max-pool 6
load 9971 sip9971.9-1-1SR1.loads
authenticate register
tftp-path flash:
create profile sync 0005135312289902
voice register dn 1
number 207
allow watch
name GossaVM
label 207
voice register dn 3
number 101
name Dejan
label 101
mwi
voice register pool 1
id mac 000C.29C5.0011
number 1 dn 1
dtmf-relay sip-notify
username testvm password testera
codec g711alaw
voice register pool 3
id mac 04C5.A4B0.3B0D
type 9971
number 3 dn 3
presence call-list
dtmf-relay rtp-nte
username dejan password 1234
codec g711alaw
no vad
license udi pid CISCO2901/K9 sn xxxxxxxxxxxx
hw-module ism 0
hw-module pvdm 0/0
redundancy
interface GigabitEthernet0/0
description INTERFACE INTERNAL
no ip address
no ip route-cache
duplex auto
speed auto
no mop enabled
interface GigabitEthernet0/0.2
description LAN DATA
encapsulation dot1Q 2
ip address 192.168.2.251 255.255.255.0
no ip route-cache
interface GigabitEthernet0/0.5
description LAN VOICE
encapsulation dot1Q 5
ip address 192.168.5.251 255.255.255.0
no ip route-cache
interface ISM0/0
no ip address
no ip route-cache
shutdown
!Application: SRSV-CUE Running on ISM
interface GigabitEthernet0/1
no ip address
no ip route-cache
shutdown
duplex auto
speed auto
interface ISM0/1
description Internal switch interface connected to Internal Service Module
shutdown
interface Vlan1
no ip address
no ip route-cache
shutdown
ip forward-protocol nd
no ip http server
no ip http secure-server
snmp-server community public RO
tftp-server flash:dkern9971.100609R2-9-1-1SR1.sebn alias dkern9971.100609R2-9-1-1SR1.sebn
tftp-server flash:kern9971.9-1-1SR1.sebn alias kern9971.9-1-1SR1.sebn
tftp-server flash:rootfs9971.9-1-1SR1.sebn alias rootfs9971.9-1-1SR1.sebn
tftp-server flash:sboot9971.031610R1-9-1-1SR1.sebn alias sboot9971.031610R1-9-1-1SR1.sebn
tftp-server flash:skern9971.022809R2-9-1-1SR1.sebn alias skern9971.022809R2-9-1-1SR1.sebn
tftp-server flash:sip9971.9-1-1SR1.loads alias sip9971.9-1-1SR1.loads
tftp-server flash:United_States/g4-tones.xml
tftp-server flash:English_United_States/gd-sip.jar
control-plane
voice-port 0/0/0
voice-port 0/0/1
voice-port 0/0/2
voice-port 0/0/3
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/1/2
voice-port 0/1/3
mgcp profile default
gatekeeper
shutdown
line con 0
line aux 0
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
password jebiga
login
transport input all
end
I did not have any kind of problem with X-LITE to register to CME. also try with few SCCP phones 7940 and I did not any kind of problem .
this is content of SEP....xml file for 9971
<device>
<deviceProtocol>SIP</deviceProtocol>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/YA</dateTemplate>
<timeZone>Pacific Standard/Daylight Time</timeZone>
<ntps>
<ntp priority="0">
<name>0.0.0.0</name>
<ntpMode>unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<sipPort>5060</sipPort>
</ports>
<processNodeName>192.168.5.251</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<localCfwdEnable>true</localCfwdEnable>
<callForwardURI>service-uri-cfwdall</callForwardURI>
<callPickupURI>service-uri-pickup</callPickupURI>
<callPickupGroupURI>service-uri-gpickup</callPickupGroupURI>
<callHoldRingback>2</callHoldRingback>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>2</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>
<sipStack>
<remotePartyID>true</remotePartyID>
</sipStack>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel></featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name></name>
<displayName></displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>dejan</authName>
<authPassword>1234</authPassword>
<sharedLine>false</sharedLine>
<messagesNumber></messagesNumber>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2" lineIndex="2">
<featureID>9</featureID>
<featureLabel>101</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>101</name>
<displayName>Dejan Rakic</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>dejan</authName>
<authPassword>1234</authPassword>
<sharedLine>false</sharedLine>
<messagesNumber></messagesNumber>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<enableVad>true</enableVad>
<preferredCodec>g711alaw</preferredCodec>
<dialTemplate></dialTemplate>
<kpml>1</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
</sipProfile>
<commonProfile>
<phonePassword>1234</phonePassword>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<featurePolicyFile>featurePolicyDefault.xml</featurePolicyFile>
<loadInformation>sip9971.9-1-1SR1.loads</loadInformation>
<vendorConfig>
</vendorConfig>
<commonConfig>
<videoCapability>0</videoCapability>
<ciscoCamera>0</ciscoCamera>
</commonConfig>
<sshUserId>dejan</sshUserId>
<sshPassword>1234</sshPassword>
<userId></userId>
<phoneServices>
<provisioning>2</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
<versionStamp>0131511014412102</versionStamp>
<userLocale>
<name>English_United_States</name>
<langCode>en</langCode>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
</networkLocaleInfo>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<servicesURL>http://192.168.5.251:80/CMEserverForPhone/serviceurl</servicesURL>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
</device>

Hello,
I'm facing exactly the same problem, that is:
a Cisco SIP Phone 9971 won't register on CME 8.6 running on a 2811
I have read all the postings to this Forum, but I have not been able to solve it.
In my case the commands voice register dn and voice register pool are OK.
So frankly, I have no idea what I could be missing.
I'm pasting the Router's config.
I hope somebody is able to point me in the right direction.
Here is the config. Thank you!
C2811#sh run
Building configuration...
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname C2811
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 172.25.140.1 172.25.140.10
ip dhcp excluded-address 172.35.140.1 172.35.140.10
ip dhcp pool Data
network 172.25.140.0 255.255.255.0
default-router 172.25.140.1
option 150 ip 172.25.140.1
dns-server 172.25.140.1
ip dhcp pool Voice
network 172.35.140.0 255.255.255.0
default-router 172.35.140.1
option 150 ip 172.35.140.1
dns-server 172.35.140.1
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 172.25.140.1 port 5060
max-dn 40
max-pool 42
load 9971 sip9971.9-4-1-9.loads
authenticate register
authenticate realm cisco
tftp-path flash:
create profile sync 0004820400584603
voice register dn 1
number 1010
allow watch
name Phone10
label Phone10
mwi
voice register pool 1
id mac 189C.5DB6.BD09
type 9971
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username adm password adm
call-forward b2bua busy 68600
codec g711ulaw
no vad
camera
video
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1879153754
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1879153754
revocation-check none
rsakeypair TP-self-signed-1879153754
crypto pki certificate chain TP-self-signed-1879153754
certificate self-signed 01
(details ommited)
license udi pid CISCO2811 sn FTX1146A44H
username admin privilege 15 password 0 admin
redundancy
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.25
description Data VLAN
encapsulation dot1Q 25
ip address 172.25.140.1 255.255.255.0
interface FastEthernet0/0.35
description Voice VLAN
encapsulation dot1Q 35
ip address 172.35.140.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 600 life 86400 requests 10000
tftp-server flash:P00308010200.bin
tftp-server flash:P00308010200.sbn
tftp-server flash:P00308010200.sb2
tftp-server flash:P00308010200.loads
tftp-server flash:SCCP42.9-3-1SR3-1S.loads
tftp-server flash:apps42.9-3-1ES19.sbn
tftp-server flash:cnu42.9-3-1ES19.sbn
tftp-server flash:cvm42sccp.9-3-1ES19.sbn
tftp-server flash:dsp42.9-3-1ES19.sbn
tftp-server flash:jar42sccp.9-3-1ES19.sbn
tftp-server flash:term42.default.loads
tftp-server flash:term62.default.loads
tftp-server flash:SCCP45.9-3-1SR3-1S.loads
tftp-server flash:apps45.9-3-1ES19.sbn
tftp-server flash:cnu45.9-3-1ES19.sbn
tftp-server flash:cvm45sccp.9-3-1ES19.sbn
tftp-server flash:dsp45.9-3-1ES19.sbn
tftp-server flash:jar45sccp.9-3-1ES19.sbn
tftp-server flash:term45.default.loads
tftp-server flash:term65.default.loads
tftp-server flash:/Ringtones/Ringlist.xml alias Ringlist.xml
tftp-server flash:/Ringtones/DistinctiveRingList.xml alias DistinctiveRingList.x
ml
tftp-server flash:sip9971.9-4-1-9.loads
tftp-server flash:kern9971.9-4-1-9.sebn
tftp-server flash:rootfs9971.9-4-1-9.sebn
tftp-server flash:dkern9971.100609R2-9-4-1-9.sebn
tftp-server flash:sboot9971.031610R1-9-4-1-9.sebn
tftp-server flash:skern9971.022809R2-9-4-1-9.sebn
tftp-server flash:/g4-tones.xml alias United_States/g4-tones.xml
tftp-server flash:/gd-sip.jar alias English_United_States/gd-sip.jar
control-plane
mgcp profile default
telephony-service
max-ephones 24
max-dn 48
ip source-address 172.25.140.1 port 2000
cnf-file location flash:
load 7960-7940 P00308010200
load 7942 SCCP42.9-3-1SR3-1S.loads
load 7945 SCCP45.9-3-1SR3-1S.loads
load 7962 SCCP42.9-3-1SR3-1S.loads
load 7965 SCCP45.9-3-1SR3-1S.loads
max-conferences 8 gain -6
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 11 2014 07:18:32
ephone-dn 1
number 1001
description Phone 1
name Phone 1
hold-alert 30 originator
ephone-dn 2
number 1002
description Phone 2
name Phone 2
hold-alert 30 originator
ephone-dn 3
number 1003
description Phone 3
name Phone 3
hold-alert 30 originator
ephone 1
device-security-mode none
mac-address 001C.58FB.6E0F
button 1:1
ephone 2
device-security-mode none
mac-address 0014.A981.7F8A
button 1:2
ephone 3
device-security-mode none
mac-address 0006.5356.A4B8
button 1:3
alias exec con conf t
alias exec sib show ip int brief
alias exec srb show run | b
alias exec sri show run int
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
line vty 5 15
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
scheduler allocate 20000 1000
ntp master 1
end
C2811#

RTP Media Stream with "Third Party SIP Device" always through CUCM

Hello,
i have i quite strage problem on one of my customers locations:
we have a cucm 7.1.5(SU4) with the cucm in the datacenter. And we have a small location(branch office) which has a small wan connection to the datacenter (1MBit/s).
In this location we have several Kirk (Polycom) Dect phones which register as "Third Party SIP Device - Basic" on the CUCM.
(The problem is the same if i use the X-Lite SIP Client instead)
When this SIP Phones or the X-Lite Client dials a internal Number of the same location the RTP Media Streams goes directly from the SIP Client to the phone. But if they dial an external number the RTP Stream goes from the SIP Client via the wan connection to the CUCM and back via the wan connection to the 2901 H.323 Gateway (on the same location).
and of course if i now start a big download or upload i'm no longe able to complete the phone call because we have no QOS on the wan connection, because we don't want to make calls over this connection.
When i look at the Sniffer files with Wireshark i see that the CUCM sends his own ip adresse in the SDP Header for the RTP Stream to the SIP Client. And this of course is wrong because the RTP Stream should always reside on the branch office.
i tested this in my lab (CUCM 8.5) and it is the same. i used the "Standard SIP Profile" and a Basic Third Party SIP Device"
The SCCP Phones on the same location which are configured with the same Region, Location, Device Pools, Media Resources and which use the same Gateway for external calls do not have this problem.
In the Gateway configuration "MTP Required" is not activated and i tested it in my lab with some Cisco SIP Phones (9971) and they are also not affacted with this problem.
any ideas?

do you have a SIP trunk to the external devices with MTP required checked ?

CUCM Third Party SIP Phone "Time" issue

Hi Team,
we have setup with Avaya 1230 SIP Phone,
and this phone we added to CUCM using "Third Party Basic SIP Phone" option.
Once registered with Call Manager "Date and time" in SIP Phones was showing fine.
we have reset the entire device pool, after that all the Avaya 1230 SIP Phone "Time" is showing +1 hour from the normal time.
How we can reslove this issue.
CUCM Version: 9.1(2a).
SIP Phone Model: Avaya 1230

Thanks for the Suggestion Manish,
I have tried the same But its not working,
In the phone level we have the option to change the "Time Zone", The same we have changed to GMT+5:30 Indian Standard Time.
Any other suggestion....

SIP phone and SPA 3102

Hi,
I don't have an ipPBX or call signaling server. Can I register a SIP phone on the remote SPA 3102 then call the remote number. SPA 3102 is on remote site, the FXO port is connected to a phone line. My SIP phone is on local site, connected to Internet.
Thanks

yytellmey wrote:
Hi,
Can I register a SIP phone on the remote SPA 3102 then call the remote number. SPA 3102 is on remote site, the FXO port is connected to a phone line. My SIP phone is on local site, connected to Internet.
Yes you can do that. You need to know the ip address where you are calling. This is called direct ip dialing. You can call the SPA3102 and have the attached phone ring, or you can call the SPA3102 and have it dial a call out the pstn line. It all depends on what you want to do.
Initially you can get it working with whatever ip address you have at the moment. For the long term, if you don't have a static ip address you can get a symbolic address from someone like dyndns.com and then when your ip address changes you setup some means, either thru your router that supports dynamic dns, or with a pc program to keep your ip address updated at dyndns for your symbolic address.
You almost always have to forward the sip signalling port in your router to the SPA3102. You may also have to forward the spa's rtp ports in your router to the SPA3102.
There are a couple of ways to configure the SPA3102 when you want to bridge the call out the pstn line. The simple way is to just return a dial tone to the caller and then the caller enters the pstn number they want to dial. A more complicated way is to send a sip invite to the SPA3102 and have the SPA dial the number. The latter method is more reliable because you don't have to send dtmf signals over your voip link.

Using Hookflash to set up conference calls in CUCM 7 with a Sip Phone

Hi all,
Not sure if I'm going the right way about this or if this is even possible, let me start by setting the scenario:
We have Cisco Unified Call Manager 7 and have recently installed an non Cisco IP based telephone in a boardroom which we have successfully registered to the Call Manager using SIP.
We would like to be able to setup audio conference calls from this SIP phone. It has a flash (hookflash) button which I believe can be used for this very purpose (it signals and offhook/onhook/offhook condition) which can be used to instruct a voice switch that it is requesting services (like conferencing).
However I'm strugging to see how to set this up.
Can anybody offer any advice on this subject?
Regards,
Martin

SG
Just an update to the question that I presented when I started this thread.
You really never did say what type of a User Account you are using in your Premiere Elements 8.0/8.0.1 on Windows 7? But based on the information provided by several users, User Account with Administrative Rights seems to be the way to go since all of them were using it and had systems that were running well. I would have expected that based on the Premiere Elements Windows XP requirements. I have yet to have someone come forward to say that they are using Premiere Elements 8.0/8.0.1 on Windows 7 with a User Account with Standard Rights and the program is running well. Will you be that one?
Besides me switching from Windows XP to Windows 7 in the near future, what motivated this line of questioning was the report that I had seen of a user with Premiere Elements 8.0/8.0.1on Windows 7 who was reporting "permission" error messages when trying to archive a project with the Project Archiver (Archiver "Trimmed" and Copy options). Further questioning brought out that the problem extended to the AutoSave feature as well. The user claimed to be saving to an external hard drive (formatted NTFS) with about 800 MB free space. And still further questioning brought out that the error message included a free space and/or permissions error. Yes, there were extremely large file sizes involved here (way over 4 GB)
The story had a happy ending where the remedies were:
1. User Account with Administrative Rights
and the major
2. Formatting the thought to be NTFS external hard drive from FAT32 to NTFS.
ATR

Using Cisco 7961 Phones with Linksys SRW208P POE Switch

I am having a problem getting our Cisco 7961 phones to work with a Linksys SRW208P POE switch. The internet works great , but when I connect a 7961 Cisco phone it hangs at the " configuring IP " screen. Are these model phones compatible with this switch? Any Ideas. Thank You!

I'm having a similar problem with the Cisco 7941G phones using a Linksys SLM224G4PS (with PoE). Did you ever get them to work?
The 7941Gs work with the plain Linksys SR224 and using power injectors, but not with the smart switch. It has nothing to do with the PoE, I've used the power injectors with the SLM224G4PS. It wasn't a current overload problem.
I've gotten the phones to work for somewhere between 10-40 seconds, which has given me enough time to find the IP address (DHCP) then pull up the web interface of the Cisco 7941G.
The phone on the SR244 had GARP enabled and Voice VLAN enabled and the operation, the phone on the SLM224G4PS had both disabled.
The working phone on the SR244 had a operational VLAN ID of 31, the non working phone on the SLM224G4PS didn't have any indicated.
I'm guessing that the root of the problem is the Voice VLAN, but tech support tells me that out of the box the switch should pass everything.
Apparently not so much...

Does anyone have a solution for using my SIP phone with my MacBook Air

I specifically purchased a MacAir for my home office. Small, Sleek, and portable. I just purchased a Cisco SIP phone and realized that I may have to continue to use my MacBook at my desk. Any work around? Is there some kind of hub or converter I can use? Thanks.

Search the web or help doco about the Captivate feature called Return to Quiz. It will do what you are looking for here according to your description. There's even a Youtube video from a year or two back that shows how it works.

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.

Cisco CP-78XX SIP Phone Pickup Not Work on CME

Hi,
I configured some SIP phones (CP-7821, CP-7841) with pickup function. Is it the Pickup / GPickup soft keys not function as the SIP phone? If yes, then I can use the FAC to access that? And I tried the FAC std. / custom as the pickup / gpickup .. both not work ... I don't know how to use the FAC on CME? As the FAC std., if I pickup local, that I should press (**3) > call?
Ref.:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmecover.html#45535
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmefacs.html#30064
This is the configuration:
CME-SIP-Phone#sh run
Building configuration...
Current configuration : 5413 bytes
! Last configuration change at 11:06:12 UTC Fri Nov 28 2014 by mtlops
version 15.4
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
service sequence-numbers
hostname CME-SIP-Phone
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.154-2.T1.bin
boot-end-marker
! card type command needed for slot/vwic-slot 0/0
enable secret 5 $XXXXXXXXXXXXXXXXXXXXXXXX
aaa new-model
aaa authentication login default local
aaa authorization console
aaa authorization exec default local
aaa session-id common
ip cef
no ipv6 cef
multilink bundle-name authenticated
stcapp feature access-code
voice-card 0
dspfarm
dsp services dspfarm
voice service pots
voice service voip
ip address trusted list
ipv4 10.118.0.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
no h225 timeout keepalive
call preserve
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 600 min 60
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice class h323 1
h225 timeout tcp establish 3
call preserve
voice class custom-cptone ABC-Company
dualtone disconnect
frequency 425
cadence 500 500
voice register pool-type 7821
description Cisco IP Phone 7821
reference-pooltype 6921
voice register pool-type 7841
description Cisco IP Phone 7841
reference-pooltype 6941
voice register global
mode cme
source-address 10.118.0.10 port 5060
timeouts interdigit 2
max-dn 200
max-pool 100
authenticate register
authenticate realm all
timezone 42
time-format 24
date-format D/M/Y
mwi stutter
mwi reg-e164
voicemail 5000
call-feature-uri pickup http://10.118.0.10/pickup
call-feature-uri gpickup http://10.118.0.10/gpickup
tftp-path flash:
file text
create profile sync 0001170446349417
ntp-server 10.118.0.10 mode unicast
ip qos dscp af11 media
ip qos dscp cs2 signal
ip qos dscp af43 video
ip qos dscp 25 service
camera
video
voice register dn 2
number 1000
pickup-call any-group
pickup-group 1
name BB Leung
label BB Leung
voice register dn 3
number 1001
pickup-call any-group
pickup-group 1
name CC Chan
label CC Chan
voice register dn 4
number 1002
pickup-call any-group
pickup-group 1
name DD Leung
label DD Leung
voice register dn 50
mwi
voice register template 1
softkeys hold Newcall Resume
softkeys idle Newcall Redial Gpickup Pickup Cfwdall DND
softkeys seized Cfwdall Endcall Redial
softkeys connected Confrn Endcall Hold Trnsfer
voice register pool 1
busy-trigger-per-button 1
id mac A8XX.XXXX.XXXX
type 7841
number 1 dn 2
template 1
dtmf-relay sip-notify
username 1001 password 112233
codec g711ulaw
no vad
voice register pool 2
busy-trigger-per-button 1
id mac 50XX.XXXX.XXXX
type 7841
number 1 dn 3
template 1
dtmf-relay sip-notify
username 1002 password 112233
codec g711ulaw
no vad
voice register pool 3
busy-trigger-per-button 1
id mac 00XX.XXXX.XXXX
type 7821
number 1 dn 4
template 1
dtmf-relay sip-notify
username 1003 password 112233
codec g711ulaw
no vad
license udi pid CISCO2921/K9 sn FHK1407F25D
license accept end user agreement
license boot c2900 technology-package uck9
hw-module pvdm 0/0
hw-module sm 1
username mtlops privilege 15 secret 5 $1$0qqx$1WGdfRW.flJrwmY7k8eUy0
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
ip address 10.118.0.10 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
interface SM1/0
no ip address
shutdown
service-module fail-open
interface SM1/1
no ip address
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 10.118.0.1
control-plane
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 7
associate application SCCP
shutdown
gatekeeper
shutdown
telephony-service
max-conferences 8 gain -6
transfer-system full-consult
fac standard
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
transport input all
scheduler allocate 20000 1000
end
CME-SIP-Phone#sh telephony-service fac
telephony-service fac standard
callfwd all **1
callfwd cancel **2
pickup local **3
pickup group **4
pickup direct **5
park **6
dnd **7
redial **8
voicemail **9
ephone-hunt join *3
ephone-hunt cancel #3
ephone-hunt hlog *4
ephone-hunt hlog-phone *5
trnsfvm *6
dpark-retrieval *0
cancel call waiting *1

VPN is not Configured prints on all phones now with the built-in VPN client if VPN isn't configured. That's normal and is just cosmetic. That should not be causing your registration issues.

CME SIP Phone Calls in one-way (inside local network)

Hello everyone, first time here, need a little help.
I'm having some trouble to find a solution to the following problem.
Recently I've installed CME 9.1 using the router 2921. Most of the phones are SIPs, model 3905 (around 20 of them), with the last firmware updated.
Some users are complaining one way audio issue in internal calls, from a extension to another (only in sip phones)
With Wireshark capture I could see that RTP packets are being sent and receive by the router and not directly trough the phones. Is this normal in CME? When a call with problems occours (one way audio) there is no audio in one way, but router still sends confort noise packets.
Here is my config.
Thanks for any help.
Martin
##################################################################################33
System returned to ROM by power-on
System restarted at 11:29:23 BR Tue Jan 29 2013
System image file is "flash0:c2900-universalk9-mz.SPA.152-4.M2.bin"
Last reload type: Normal Reload
Last reload reason: power-on
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 10.3.245.1 port 5060
max-dn 60
max-pool 70
load ATA-187 ATA187.9-2-3-1
load 3905 CP3905.9-2-1-0
authenticate realm all
timezone 17
time-format 24
date-format D/M/Y
tftp-path flash:
file text
create profile sync 0094230880392697
network-locale U1
user-locale U1 load /CME-locale-pt_BR-Portuguese-8.8.2.5.tar
ntp-server 10.3.244.7 mode directedbroadcast
voice register dn 1
number 9006
name Sala_Reuniao_02
label Sala de Reuniao 2
voice register dn 2
number 9007
name Sala_Reuniao_03
voice register dn 3
number 9008
name Sala Reuniao 04
voice register pool 1
id mac 8478.ACE6.09A2
type 3905
number 1 dn 1
template 1
codec g711ulaw
voice register pool 2
id mac 8478.ACE6.0573
type 3905
number 1 dn 2
codec g711ulaw
voice register pool 3
id mac 5897.1ECD.8F8D
type 3905
number 1 dn 3
codec g711ulaw
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.220
encapsulation dot1Q 220
ip address 10.3.245.1 255.255.255.0
ip helper-address 10.3.244.71
h323-gateway voip bind srcaddr 10.3.245.1
telephony-service
max-ephones 5
max-dn 5 no-reg both
ip source-address 10.3.245.1 port 2000
timeouts interdigit 5
timeouts busy 12
system message XXXXXXXX
cnf-file location flash:
cnf-file perphone
user-locale U2 load CME-locale-pt_BR-Portuguese-8.8.2.5.tar
user-locale 2 PT
network-locale U2
load 7925 CP7925G-1.4.1SR1.LOADS
load 6941 SCCP69xx.9-2-1-0.loads
time-zone 17
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 9001
ephone 1
mac-address D867.D9E6.F57F
ephone-template 1
type 6941
button 1:1

Hi ,
We have upgarded the the firmware to the 3905.9-2-2ES2 , but show voice register pool phone-load still shows the old firmware, but the phoen itself is showing the new upgraded version on the dsiplay ...any advice is highly appricated,
ADM-CME9#show voice register pool phone-load
Pool Device Name     Current-Version             Previous-Version
==== =============== =========================== ===========================
1    SEP7081053DE72F Cisco/SPA502G-7.4.8a
3    SEP34BDC8C6C412 Cisco-CP3905/9.2.1
4    SEP34BDC8C64561 Cisco-CP3905/9.2.1
5    SEP54781AE1F531 Cisco-CP3905/9.2.1
6    SEP54781AE171D2 Cisco-CP3905/9.2.1
10   SEP54781AE1F544 Cisco-CP3905/9.2.1
15   SEP1CE6C77323CD Cisco-CP3905/9.2.1
16   SEP58971E282A23 Cisco-CP3905/9.2.1
17   SEP58971E2822A8 Cisco-CP3905/9.2.1
19   SEP1CE6C77321F3 Cisco-CP3905/9.2.1
30   SEP54781AE171E2 Cisco-CP3905/9.2.1
31   SEP54781AE16FD4 Cisco-CP3905/9.2.1
32   SEP54781AE16F2F Cisco-CP3905/9.2.1
33   SEP54781A1C77FD Cisco-CP3905/9.2.1
34   SEP54781A1C77DC Cisco-CP3905/9.2.1
35   SEP54781AE17527 Cisco-CP3905/9.2.1
36   SEP54781AE17766 Cisco-CP3905/9.2.1
37   SEP54781AE1731A Cisco-CP3905/9.2.1
38   SEP54781AE08B8D Cisco-CP3905/9.2.1
39   SEP54781AE123B1 Cisco-CP3905/9.2.1

SPA IP phones doesn't show calling number after the call was picked UP

Hello,
Our different CISCO SPA IP phones models (at this time we are using 7.4.8a version) are working with Broadsoft AS. And now we have a problem: when CISCO SPA IP PHONE pickups the call ( by pressing BLF lamp) and the call is connected, Cisco user doesn't see any information (calling name or calling number) with whom he is talking now. He on only sees on screen pickup extension *11 and info that the call is connected. From SIP traces I can see this information in PAI header in SIP 200 OK message (when pickup was initiated).
I also tried to change Caller ID Header - no luck. I've upgraded IP phone to latest version (7.5.1a) - didn't help.
I've experience with THOMSON and MGC. By pressing BLF , Thomson displays info who (calling number) is calling to your configured BLF line and gives you actions: retrieve this call or cancel - you can choose if you still want pickup this call.
So can you give me some hints how to fix this issue? Our at this time CISCO IP PHONE doesn't have such feature?
Thanks in advance,
Giedrius

Why not answer?
bug is available from firmware 7-4-7 and above and are still
on models SPA50x and SPA525

Configure SPA2102 as SIP Phone to SIP Proxy Server

I have a SIP2102 which needs to be configured as SIP phone to a SIP Proxy Server. All calls will stay within the local network. I need to point SPA2102 to my SIP Proxy Server and assign an extension such that it is recognized by the SIP Proxy Server as part of its pool of valid extensions. The documentation is not clear when just trying to set these parameters.

if what you are trying to accomplish is simply register the SPA2102 with your Proxy server then the only thing you need to do is configure a USER ID (this is the extension number you want the SPA to have) and a Proxy (IP address of your SIP server) -- outbound proxy is only needed if your server requires the device to have this..
Both of these parameters can be found under Line 1 tab and these settings should come from your SIP server-- Internet port of the device must be connected to your network
SIP Port is another to consider in case your SIP server is using another port other than 6060..
| isolate! isolate! isolate! |

CCME with Linksys SPA SIP phones

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