Forwarded calls are not "call-parked"
Dear Lync experts,
please could You help me that the following scenario is not possible or I need to change some config parameter:
- there are two extensions ("A" and "B")
- a call park group was defined and both extension were added to this call-park group (call pickup group)
- extension "A" was forwarded to extension "B" (using Lync client- Call Forwarding - Forward my call to- Delegate "B")
- there is an incoming call to "A", it is ringing at "B" (because of the forward option), but dialing the call-park-group number we are not able to pickup this call.... I do not understand why?!
Thanks in advance:
Norbert
Hi,
Would you please tell us why you want to forward call to another account in the same call park group?
Base on my understanding, if cancel the call forwarding, other users could pick up the call. Please check if it works after cancelling call forwarding setting. Please check if the issue still happen when cancelling call forwarding.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support
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Hi,
Am trying to develop a report the parameter has an issue, it gives me an error.
There are 2 parameter (1st is Date range like start and end date and 2nd is Loan number)
When ever I select date range the Loan number parameter should go blank. If I input a loan number then date range should go blank.
What should I do. Please help
ERROR --report parameter startdate has a defaultvalue or a valid value that depends on the report parameter startdate forward dependencies are not validHi Kay_MD,
If there are only two parameters, we can use first parameter charge the second one, but the second one cannot charge the first one in turn. Because parameter forward dependencies are not valid. To achieve your requirement, we can create another parameter
to control Date_range and Loan_number parameters. For more details, please see:
Add a parameter named ReportParameter1 with two available values: Date_range and Loan_number.
Use the following expression as the default value in Date_range parameter:
=iif(Parameters!ReportParameter.Value="Date_range",1,"") (1 means a useless value)
Use the following expression as the default value in Loan_number parameter:
=iif(Parameters!ReportParameter.Value="Loan_number",1,"") (1 means a useless value)
Please note that the Date_range and Loan_number parameters should enable Allow blank value (“”), and the blank values should also be contained in available values.
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Thanks,
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Katherine Xiong
TechNet Community Support -
Forwarding Rates are not accurate
Hi,
Yesterday I activated forwarding calls, and I have two questions :
Rates to United Stats is 2.3¢/min
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https://dl.dropboxusercontent.com/u/13684246/Skype2.jpg
The second question is :
couples of my friends reported that after skype starts ringing for few seconds, It doesn't forward their call to my phone properly !!! It stops ringing when It supposedly forwarding their call to me !!!Hi,
According to your post, my understanding is that search results are not accurate.
I recommend to run a full crawl and then to check whether the search results are correct.
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Linda Li
Linda Li
TechNet Community Support -
When I am surfing the net, I have always been able to use my back/forward buttons. Now all of a sudden, they are not available to use, you can still see them, but they are grey, not black, like they once were so that I could click them to go back to a previous page, or forward once I've went back? I don't know what's the problem. I did the most recent firefox update that was sent to me recently. But just noticed this problem today. Thanks for your help.
A possible cause is a problem with the file places.sqlite that stores the bookmarks and the history.
*http://kb.mozillazine.org/Bookmarks_history_and_toolbar_buttons_not_working_-_Firefox
Start Firefox in <u>[[Safe Mode]]</u> to check if one of the extensions is causing the problem (switch to the DEFAULT theme: Firefox (Tools) > Add-ons > Appearance/Themes).
*Don't make any changes on the Safe mode start window.
*https://support.mozilla.com/kb/Safe+Mode -
Close, new, back, forward icons are not shown
Close, new, back, forward icons in firefox are not shown, so it makes difficult to use firefox. I am using latest firefox in ubuntu 11.04.
Already asked here but no response :( superuser.com/questions/559381/in-firefox-back-tab-close-new-tab-icon-is-not-shown I have posted screenshot there, It will be helpful so posting the link here.The Reset Firefox feature can fix many issues by restoring Firefox to its factory default state while saving your essential information.
Note: ''This will cause you to lose any Extensions, Open websites, and some Preferences.''
To Reset Firefox do the following:
#Go to Firefox > Help > Troubleshooting Information.
#Click the "Reset Firefox" button.
#Firefox will close and reset. After Firefox is done, it will show a window with the information that is imported. Click Finish.
#Firefox will open with all factory defaults applied.
Further information can be found in the [[Reset Firefox – easily fix most problems]] article.
Did this fix your problems? Please report back to us! -
Rule Forwarded Messages are NOT in Sent folder
I have several rules which filter & automatically forward messages elsewhere. The forward works most of the time BUT it does not appear in my Sent folder. Any insight or help would be appreciated...
Robyn
Message was edited by: derryhummaThanks for that hint. Unfortunately I was not so lucky as you were. Also when trashing the .plist only the mac.com account remains in the account list after restart of Apple Mail. All other accounts need to be reinstalled
Still Mail does not store the mails in the sent folder. Need to say:
- the "non-storing" is random, not all sent mails are affected. Sometime it is OK, sometimes not
- Not only the Mac account is affected, it can happen also on my office accounts.
I am happy for further hints -
anyone can you help my earphones are not working on my iphone4 and i know its not the earphones has just bought new pair and no joy I can not play, stop, forward to different tracks or anything cant answer call nothing end of ideas can anyone help pretty please
Yes there genuine apple earphones I have now blown down jack slot and silly enough I turned my phone off and now it's back on earphones working properly now feeling a wee but silly as such a simple thing got them working.
Thank you guys and I know where to come again if I need any further assistance.
Again thank you everyone -
Calls are not getting thru in Cisco voice GW for a particular Number
Cisco gateway is connecte to a PBX with an Qsig interface, for a particualr destination number the calls are not gettin estabilished.
the output of the Q931 debug :
Aug 16 16:17:46.145: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x7E05
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98396
Exclusive, Channel 22
Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
Facility i = 0x9FAA068001008201008B0100A11D0202010002010080144E455453202
F204C4F4E472044495354414E4345
Calling Party Number i = 0x2183, '8168911010'
Plan:ISDN, Type:National
Called Party Number i = 0x89, '18553808521'
Plan:Private, Type:Unknown
Sending Complete
Aug 16 16:17:46.149: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0xF
E05
Channel ID i = 0xA98396
Exclusive, Channel 22
Aug 16 16:17:55.709: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x
FE05
Cause i = 0x80BF - Service/option not available, unspecified
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x7E0
5
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref =
0xFE05
The Qsig and dial-peer configration :
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vadHi Raj,
My name is Edson Pineiro, I understand that your problem description is in regards to failed incoming calls from a qsig trunk.
According to the received q931 setup message I can see the called party number is 18553808521 and as so the gateway should route the dnis based on the best match in destination-pattern. My first suggestion would be to ensure your outgoing dial-peers has a matching destination-pattern that matches the dialed number, for example:
dial-peer voice 1 voip
destination-pattern 1T
The T is a wild card for any digit any length
Or you can be very specific.
dial-peer voice 1 voip
destinaton-pattern 18553808521
The next suggestion would be to ensure that your incoming pots dial-peers contains 'direct-inward dial'. This is so that you don't receive secondary dial tone when dialing in, which I don't think is happening here.
Another suggestion would be to remove 'isdn overlap-receiving' from interface serial 0/0/0:23. Reason being is that the DNIS received is enbloc and not overlapping. You can clearly see that the complete e164 number is received within the setup and no further digits are needed.
But overall the disconnect cause code is 0x80BF the 80 portion is related to the source of the disconnect which is the router and BF "Service/option not available, unspecified" which is described as:
The network or remote equipment cannot provide the service option that the user requests, due to an unspecified reason. A subscription problem can cause this issue.
Any ways seems like the router does not support the protocol or type of message included in the Setup. After decoding one of the facility message:
Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
decode -->
Facility IE first byte (protocol profile): 0x9f(Network Extentions), depends on Network Protocol Profile
**Note:
**0x91/0x9f both be used by older qsig spec, including:
**ISO 11582:1995, ETSI 300 239:1993/1995
**newer qsig spec use 0x9f only, including:
**ISO 11582:1995/Cor.1:1999, ECMA 165(4th), ETSI 300 239:2003
**see CSCeb58118 for CCM compatibility issue
NetworkFacilityExtension ::= {
sourceEntity: 0
destinationEntity: 0
NetworkProtocolProfile not present
APDU is a ROSE
0
DivertingLegInformation2Invoke ::= {
invokeID: 1793
operationValue: 21
argument: DivertingLegInformation2Arg ::= {
diversionCounter: 1
diversionReason: 1
originalDiversionReason: 1
divertingNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
originalCalledNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
redirectingName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
originalCalledName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
Looks like this is a redirected call (call forward or transfer), the redireted number is "50005998" and the other end of the PRI maybe attempting to do either a 2 B channel transfer or B channel optimization, which is not supported certain gateways or needs the use of a tcl scripts. Any ways is it possible to confirm if such features are enabled on the other end of the qsig trunk? and what the number 50005998 is assigned too. This may warrant a TAC case.
However please ensure your carry through the first three configuration changes before looking at the possible bad facility message.
Here are some good documents on ISDN, IOS dial-peers and call legs:
Understanding debug isdn q931 Disconnect Cause Codes
http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml
Configuring Telephony Call-Redirect Features
Two B-Channel Transfer
http://www.cisco.com/en/US/docs/ios/voice/ivr/pre12.3_14_t/configuration/guide/ivrapp.pdf
Understanding Dial Peers and Call Legs on Cisco IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
Understanding Direct-Inward-Dial (DID) on IOS Voice Digital (T1/E1) Interfaces
http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_note09186a00801142f8.shtml
Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#prereq
Voice Translation Rules
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
Let me know how you go.
Thanks again for asking the tuff questions.
Cheers
Edson -
Issue with SPA525g registation and FXO port call calls are not disconnecting properly
Hi,
I have a UC540 and updated it to the latest IOS version with the latest firmware to my phones and i am having registration problems with SPA525g IP Phones. I updated the firmware of the phones as well and create manual tftp bindings with but still it is not registering. I run a couple of debugs (debug tftp events and debug ephone registration) I can see from the logs and in the phone that it is taking the proper VLAN and being discovered via CDP and being pointed to the TFTP server and still wont register. I can see that it is also taking its own .cnf file properly then the output sccp token regected invalid devices error is shown I have a SPA502G and it is working fine. Also there is a previous issue that all the voice port are shown as engage or offhook even the calls are disconnected thus make the main PSTN number busy am based in UAE and our service provider is etisalat I have check with them about the proper disconnection values but still it the same. That's why I have arrived in the conclusion to just update everything including the IOS and the phones firmware. I have put my config in this post, I am also trying to take the CCNA Voice exam on the 2nd week of april and I think that if i don't know how fix this issue for our customer then I would probably fail that exam. any suggestion and help is greatly appreciated cisco experts.
! Last configuration change at 13:36:42 ZP4 Thu Sep 13 2012 by Nick
! NVRAM config last updated at 13:45:41 ZP4 Thu Sep 13 2012 by Nick
version 15.1
parser config cache interface
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
service compress-config
service sequence-numbers
hostname UC540
boot-start-marker
boot system flash:uc500-advipservicesk9-mz.151-2.T4
boot-end-marker
logging buffered 64000
enable secret 5 $1$3CIf$.rXyHeJQrwd97X/f2dS0M1
no aaa new-model
clock timezone ZP4 4 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-3558175224
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-3558175224
revocation-check none
crypto pki certificate chain TP-self-signed-3558175224
certificate self-signed 01 nvram:IOS-Self-Sig#3.cer
dot11 syslog
dot11 ssid cisco-data
vlan 1
authentication open
dot11 ssid cisco-voice
vlan 100
authentication open
ip source-route
ip cef
ip dhcp relay information trust-all
ip dhcp excluded-address 10.1.3.1 10.1.3.10
ip dhcp pool phone
network 10.1.3.0 255.255.255.0
default-router 10.1.3.1
option 150 ip 10.1.3.1
ip name-server 213.42.20.20
ip name-server 195.229.241.222
ip inspect WAAS flush-timeout 10
ip inspect name SDM_LOW cuseeme
ip inspect name SDM_LOW dns
ip inspect name SDM_LOW ftp
ip inspect name SDM_LOW h323
ip inspect name SDM_LOW https
ip inspect name SDM_LOW icmp
ip inspect name SDM_LOW imap
ip inspect name SDM_LOW pop3
ip inspect name SDM_LOW netshow
ip inspect name SDM_LOW rcmd
ip inspect name SDM_LOW realaudio
ip inspect name SDM_LOW rtsp
ip inspect name SDM_LOW esmtp
ip inspect name SDM_LOW sqlnet
ip inspect name SDM_LOW streamworks
ip inspect name SDM_LOW tftp
ip inspect name SDM_LOW tcp router-traffic
ip inspect name SDM_LOW udp router-traffic
ip inspect name SDM_LOW vdolive
no ipv6 cef
multilink bundle-name authenticated
stcapp ccm-group 1
stcapp
stcapp supplementary-services
port 0/0/0
fallback-dn 301
port 0/0/1
fallback-dn 302
port 0/0/2
fallback-dn 303
port 0/0/3
fallback-dn 304
trunk group ALL_FXO
max-retry 5
voice-class cause-code 1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_FXO
trunk group ALL_FX0
voice call send-alert
voice rtp send-recv
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
sip
no update-callerid
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
voice class dualtone-detect-params 1
freq-max-deviation 50
freq-max-power 0
freq-min-power 13
freq-power-twist 4
cadence-variation 6
voice class custom-cptone UAE-CUSTOM
dualtone disconnect
frequency 406
cadence 398 344 237 527 400
voice class custom-cptone CCAjointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50
voice class custom-cptone CCAleavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
voice class cause-code 1
no-circuit
voice register global
voice hunt-group 1 parallel
list 301,302,303
timeout 24
pilot 511
voice translation-rule 4
rule 15 // //
voice translation-rule 1000
rule 1 /.*/ //
voice translation-rule 1111
voice translation-rule 1112
rule 1 /^9/ //
rule 3 /^0/ //
voice translation-rule 2222
voice translation-rule 3265
rule 1 /\(^..........$\)/ /9\1/
rule 2 /\(^.........$\)/ /9\1/
rule 15 /\(^ABCD$\)/ /ABCD\1/
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile INCOMING_CallerID_PROFILE
translate calling 3265
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
voice translation-profile PROFILE_ALL_FXO
translate calling 4
voice translation-profile nondialable
translate called 1000
voice-card 0
dspfarm
dsp services dspfarm
license udi pid UC540W-FXO-K9 sn FHK143074G6
archive
log config
logging enable
logging size 600
hidekeys
username cisco privilege 15 secret 5 $1$vjNa$OFKLhupqR8al6x2b8Xmcj/
username adminac privilege 15 secret 5 $1$NDC.$PtD0y4YGIj5SqI1gghxWE1
username Nick privilege 15 secret 5 $1$iAmL$tsg7Jf2TEND1NN.h8z2dy/
ip tftp source-interface Loopback0
bridge irb
interface Loopback0
description $FW_INSIDE$
ip address 10.1.10.2 255.255.255.252
ip access-group 101 in
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/0
description $FW_OUTSIDE$
ip address 192.168.101.2 255.255.255.252
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
interface Integrated-Service-Engine0/0
description cue is initialized with default IMAP group
ip unnumbered Loopback0
ip nat inside
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface FastEthernet0/1/0
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/1
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/2
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/3
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/4
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/5
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/6
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/7
switchport access vlan 20
spanning-tree portfast
interface FastEthernet0/1/8
switchport access vlan 100
macro description cisco-switch
interface Dot11Radio0/5/0
no ip address
shutdown
ssid cisco-data
ssid cisco-voice
speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
station-role root
interface Dot11Radio0/5/0.1
encapsulation dot1Q 1 native
bridge-group 1
bridge-group 1 subscriber-loop-control
bridge-group 1 spanning-disabled
bridge-group 1 block-unknown-source
no bridge-group 1 source-learning
no bridge-group 1 unicast-flooding
interface Dot11Radio0/5/0.100
encapsulation dot1Q 100
bridge-group 100
bridge-group 100 subscriber-loop-control
bridge-group 100 spanning-disabled
bridge-group 100 block-unknown-source
no bridge-group 100 source-learning
no bridge-group 100 unicast-flooding
interface Vlan1
no ip address
bridge-group 1
bridge-group 1 spanning-disabled
interface Vlan20
ip address 10.10.10.1 255.255.255.0
interface Vlan100
no ip address
bridge-group 100
bridge-group 100 spanning-disabled
interface BVI1
description $FW_INSIDE$
no ip address
ip nat inside
ip virtual-reassembly in
shutdown
interface BVI100
description $FW_INSIDE$
ip address 10.1.3.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:/gui
ip dns server
ip nat inside source list 1 interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 192.168.101.1
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
logging esm config
access-list 1 remark SDM_ACL Category=2
access-list 1 permit 192.168.10.0 0.0.0.255
access-list 1 permit 10.1.3.0 0.0.0.255
access-list 1 permit 10.1.10.0 0.0.0.3
access-list 100 remark auto generated by SDM firewall configuration
access-list 100 remark SDM_ACL Category=1
access-list 100 deny ip 192.168.10.0 0.0.0.255 any
access-list 100 deny ip host 255.255.255.255 any
access-list 100 deny ip 127.0.0.0 0.255.255.255 any
access-list 100 permit ip any any
access-list 101 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 101 remark SDM_ACL Category=1
access-list 101 permit tcp 10.1.3.0 0.0.0.255 eq 2000 any
access-list 101 permit udp 10.1.3.0 0.0.0.255 eq 2000 any
access-list 101 deny ip 10.1.3.0 0.0.0.255 any
access-list 101 deny ip 192.168.10.0 0.0.0.255 any
access-list 101 deny ip 192.168.101.0 0.0.0.3 any
access-list 101 deny ip host 255.255.255.255 any
access-list 101 deny ip 127.0.0.0 0.255.255.255 any
access-list 101 permit ip any any
access-list 102 remark auto generated by SDM firewall configuration##NO_ACES_6##
access-list 102 remark SDM_ACL Category=1
access-list 102 deny ip 10.1.10.0 0.0.0.3 any
access-list 102 deny ip 10.1.3.0 0.0.0.255 any
access-list 102 deny ip 192.168.101.0 0.0.0.3 any
access-list 102 deny ip host 255.255.255.255 any
access-list 102 deny ip 127.0.0.0 0.255.255.255 any
access-list 102 permit ip any any
access-list 102 permit ip 192.168.101.0 0.0.0.3 any
access-list 103 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 103 remark SDM_ACL Category=1
access-list 103 permit tcp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 deny ip 10.1.10.0 0.0.0.3 any
access-list 103 deny ip 192.168.10.0 0.0.0.255 any
access-list 103 deny ip 192.168.101.0 0.0.0.3 any
access-list 103 deny ip host 255.255.255.255 any
access-list 103 deny ip 127.0.0.0 0.255.255.255 any
access-list 103 permit ip any any
access-list 105 permit ip any any
snmp-server community public RO
tftp-server flash:/phones/521_524/cp524g-8-1-17.bin alias cp524g-8-1-17.bin
tftp-server flash:/phones/5x5/spa5x5-7-1-3c.bin alias spa5x5-7-1-3c.bin
tftp-server flash:/phones/525/spa525g-7-4-8.bin alias spa525g-7-4-8.bin
control-plane
bridge 1 route ip
bridge 100 route ip
voice-port 0/0/0
cptone GB
station-id name Cordless
station-id number 329
caller-id enable
voice-port 0/0/1
cptone AE
caller-id enable
voice-port 0/0/2
cptone AE
caller-id enable
voice-port 0/0/3
cptone AE
caller-id enable
voice-port 0/1/0
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4FXO-0/1/0-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/1
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/1-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/2
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
supervisory dualtone-detect-params 1
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/2-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/3
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/3-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music On Hold Port
sccp local Loopback0
sccp ccm 10.1.3.1 identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register confprof1
dspfarm profile 1 conference
description DO NOT MODIFY, active CCA conference profile - CCA2.0 codec729
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dial-peer cor custom
name internal
name local
name local-plus
name international
name national
name national-plus
name emergency
name toll-free
dial-peer cor list call-internal
member internal
dial-peer cor list call-local
member local
dial-peer cor list call-local-plus
member local-plus
dial-peer cor list call-national
member national
dial-peer cor list call-national-plus
member national-plus
dial-peer cor list call-international
member international
dial-peer cor list call-emergency
member emergency
dial-peer cor list call-toll-free
member toll-free
dial-peer cor list user-internal
member internal
member emergency
dial-peer cor list user-local
member internal
member local
member emergency
member toll-free
dial-peer cor list user-local-plus
member internal
member local
member local-plus
member emergency
member toll-free
dial-peer cor list user-national
member internal
member local
member local-plus
member national
member emergency
member toll-free
dial-peer cor list user-national-plus
member internal
member local
member local-plus
member national
member national-plus
member emergency
member toll-free
dial-peer cor list user-international
member internal
member local
member local-plus
member international
member national
member national-plus
member emergency
member toll-free
dial-peer voice 1 pots
port 0/0/0
no sip-register
dial-peer voice 2 pots
port 0/0/1
no sip-register
dial-peer voice 3 pots
port 0/0/2
no sip-register
dial-peer voice 4 pots
port 0/0/3
no sip-register
dial-peer voice 5 pots
description ** MOH Port **
destination-pattern ABC
port 0/4/0
no sip-register
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/1
dial-peer voice 52 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/2
dial-peer voice 53 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/3
dial-peer voice 54 pots
description ** FXO pots dial-peer **
destination-pattern A0
port 0/1/0
no sip-register
dial-peer voice 55 pots
description ** FXO pots dial-peer **
destination-pattern A1
port 0/1/1
no sip-register
dial-peer voice 56 pots
description ** FXO pots dial-peer **
destination-pattern A2
port 0/1/2
no sip-register
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 388
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 6 pots
description "catch all dial peer for BRI/PRI"
translation-profile incoming nondialable
incoming called-number .%
direct-inward-dial
dial-peer voice 57 pots
description ** FXO pots dial-peer **
destination-pattern A3
port 0/1/3
no sip-register
dial-peer voice 69 pots
destination-pattern 329
port 0/0/0
dial-peer voice 300 pots
trunkgroup ALL_FX0
description Local Numbers
destination-pattern 9T
forward-digits 9
dial-peer voice 301 voip
destination-pattern 2..
session target ipv4:192.168.201.2
dial-peer voice 303 pots
trunkgroup ALL_FXO
trunkgroup ALL_FX0
description **InternationalCall**
destination-pattern 88T
dial-peer voice 304 pots
trunkgroup ALL_FX0
description *EM1*
destination-pattern 9[1-9]T
forward-digits 3
dial-peer voice 302 pots
trunkgroup ALL_FX0
description **Mobiles**
destination-pattern 9.[0-9].[0-9]......
dial-peer voice 305 pots
trunkgroup ALL_FX0
description **800-**
destination-pattern 9[0-9][0-9][0-9]T
no dial-peer outbound status-check pots
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 5
sdspfarm tag 1 confprof1
conference hardware
video
fxo hook-flash
max-ephones 40
max-dn 300
ip source-address 10.1.3.1 port 2000
max-redirect 20
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service phone webAccess 0
service dnis overlay
service dnis dir-lookup
timeouts interdigit 5
system message American Center
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.2/CCMCIP/authenticate.asp
load 521G-524G cp524g-8-1-17
load 525G spa525g-7-4-8
load 501G spa5x5-7-1-3c
load 502G spa5x5-7-1-3c
load 504G spa5x5-7-1-3c
load 508G spa5x5-7-1-3c
load 509G spa5x5-7-1-3c
time-zone 35
date-format dd-mm-yy
voicemail 388
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
moh MOH2.wav
multicast moh 239.10.16.16 port 2000
web admin system name cisco secret 5 $1$iDgA$MKNi2RWfsO0KjuC82kgLJ1
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
transfer-pattern .T
secondary-dialtone 9
fac standard
create cnf-files version-stamp 7960 Aug 29 2012 12:00:04
line con 0
privilege level 15
logging synchronous
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
line vty 0 4
exec-timeout 0 0
logging synchronous
login local
transport input all
line vty 5 100
login local
transport input all
ntp master
end
Some of the output are not shown becaus it is to long I have attach the whole config for reference and any advice on how could I optimize and resolve my issues is greatly appreciated. ThanksNicolo - First off this stuff gets crazy sometimes. No worries about the exam. Sometimes when FXO ports go crazy it is due to battery reversal. If you go to the FXO port settings try turning battery reversal on and or off... depending on its current setting. See if that helps.
As for the 525s not registering.. These are inside the network correct? Are you connecting one directly to the UC500 with a Cat5E or Cat6 patch cable and the same thing happens? Does the MAC address on the phone match a MAC address under the EPHONE settings?
If you telnet into the UC500 can you execute a "dir" command at the CLI prompt and "CD" (change directory) into the phones folder and then the spa525g folder? Do files exist in there?
Also I only see an IP address under BVI100? This is the voice side of things what happened to the IP address under BVI1 (Data VLAN). Can you give us some information about the internal network? Cna you PING this phone system from the network? What IP address does it have? -
Online Number is not forwarding calls to my mobile
Hi,
I have gone premium and selected unlimited landline for Australia. I have then bought an Online Australian Number. After setting up call forward to my Indian Mobile Number the calls are not getting forwarded, although the calls appear on Skype Client when I'm online. I have tried setting privacy and all that stuff, but it yet does not work. In the Call Forward tab following message is displayed: "All calls forwarded to phones covered by your subscription are included at no extra cost. Other destinations will be charged at our low per minute rates."
Am I missing something? Do I still need to buy credit? Thanks so much,
Kind Regards
PrabhHello,
Skype Number enables you to receive calls on your Skype account and the caller will be charged local rates.
To receive calls to your Skype Number, you should be online on your Skype application, or you can enable call forwarding and voice messaging. And also, please make sure that you are signed in to only one device.
Please make sure that you have set up your Privacy Settings to "Received calls from anyone."
Sign in to Skype.
Tap the menu icon .
Select Settings.
Go to Receive calls from option.
If one of my replies has addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others experiencing the same issue. -
MOH not working in Directed call park
Hello All,
After upgrading the voice gateway from 15.1(2)T2 to15.1(3)T4 im facing this issue.
MOH is not working in Directed call park. We have 10 Directed call park numbers 100-109, users are parking the calls by pressing trasfer key (transfer key-->dial 100-->transfer key). The Music is stopped playing after the user presses the transfer key for the 2nd time. MOH is working fine in hold and call park. I can see the MOH session in Show ccm-manager music-on-hold and the same MOH session is continuing after pressing the transfer for the 2nd time, only the music is stopping. There are some sites in the same cluster which MOH working in all conditons but the router is rnning with
Using: 2900 series voice gateway with H.323, Centralized cluster.
Assistace will be musch appricidated
Thanks,
MuraliThis issue has been identified as new BUG. Cisco is working on this.
-M -
Forwarded call to cell phone is not displayed with...
I have a subscription allowing me to let calls to my skype online-number be forwarded to my cell phone. Now, if someone calls my skype number and is forwarded to my cell phone, my cell phone only shows "unknown". This is also the case, if the person calling me and being forwarded has activated his phone to show its number. So it has to be a setting inside the skype-program. How can I change this setting to let the number calling me being displayed on my cell phone?
Thank you, I appreciate your help very much.
Solved!
Go to Solution.faju wrote:
I have a subscription allowing me to let calls to my skype online-number be forwarded to my cell phone. Now, if someone calls my skype number and is forwarded to my cell phone, my cell phone only shows "unknown". This is also the case, if the person calling me and being forwarded has activated his phone to show its number. So it has to be a setting inside the skype-program. How can I change this setting to let the number calling me being displayed on my cell phone?
It is possible that the telephone network is not passing the Caller ID information along. This is the case in certain countries, even if the caller has enabled Caller ID and you have set a number in your Skype account to be used as Caller ID. Have you done that yet? Log into your account here on the Skype web site using the "Account" link at the top of this page to see your Skype account's Caller ID settings.
If you have Caller ID set up on your Skype account, sometimes you will see your own Caller ID number when calls are forwarded from Skype. Other times, it may be "unknown" or a series of digits. Unfortunately, Skype does not have control over all of this, since Skype works with partners in many countries to provide services using the telephone network (making calls to or receiving calls from telephones, forwarding calls from Skype to a telephone number, Caller ID, etc.).
Patrick
Location/Ubicacion: Arizona USA
Time Zone/Hora Local: UTC/GMT -7
If this message has adequately addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others.
Si esto mensaje le ha ayudado, por favor haga clic en "Aceptar como solución". Si encuentra un mensaje útil, por favor "Da Kudos" al final del mensaje, por lo que esta información puede beneficiar a otros.
I am not a Skype employee. No soy un empleado de Skype. -
External PSTN calls are not making it to CUE voicemail?
External PSTN calls are not being forwarded to my Unity Express Voicemail. Internall calling and forwarding to Voicemail is working like a charm. However, calls from the outside just busyout after so many rings and never make it to voicemail.
Can some one help point me in the right direction. I think it might be as simple as dial-peer problem.
Thanks in advance,
CurtI'm assuming this is CME-integrated. Basically, if you have a PRI connection, the easiest thing to do is to do a 'debug isdn q931' and check out the called number. Or if you have an FXO connection to the PSTN, then you've got some kind of 'connection plar' under the voice port to give you the called number. Basically the called number must match a destination pattern under a voip dial-peer pointing to CUE (you can use 'debug ccsip message' and look for an INVITE message to see where it's going to). Next, it has to match a trigger ('show ccn trigger' in the CUE CLI) to envoke a particular application, such as voicemail. If you have trouble, what we'll need to see is the exact called number (either via the isdn q931 debug or connection plar configuration statement) and the configuration (show run) from the CME and the CUE.
Lastly, take a look at the following notice:
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_field_notice09186a008023cfe2.shtml
which might apply to your situation. -
Hi. please solve my problem.
.call information such as call waiting,call forwarding, call holding is not working when caller the call me or im wait for my call i cant see any title by iphon , and also holding is the same.2. there is not any option for call baring.3.playback music is not hearing by second partner during the call.4.i cant select ringing ton from saving tons.
thank you in advance.Hi Ersin,
Exception 1 would seem to be FORMATTING_ERROR, which suggests something in the design of the Smartform. However, that doesn't fit with the form being generated from a report, but only erroring when called from a function module. I can think of no reason why the different calling method would be a factor.
When FORMATTING_ERROR is raise it should also set a message ID and number, are you able to determine what they are?
Regards,
Nick -
CCD/SAF learned DNs are not able to call
Hi,
I have configured SAF in CUCMEs and configured a SAF Forwarder for CUCM. All CUCMEs are learning and advertising DNs and able to call each other. SAF Forwarder registered CUCM as client and in CUCM I can see learned DNs too. CUCMEs are learning CUCM DNs too SAF. I have created partition for learned DNs and CSS include that partition. But I am unable to call to these numbers. while digit analysis these learned DNs are not matched with any partitions and shows blocked due to unallocated number. Can someone help me to figure out what I missed in this configuration?
Thanks.I tried , but no luck.
In DNA, I cannot see the specific DNs are matching to any partition. But i have partition already for CCD learned DNs and I put that in Calling Search Space. In DNA it is unallocated Number. As you said I made feature configuration but still it is same.
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