Voice router

i am using cisco 3745 router for voice router i am facing a problem after few days all calls are blocking once it restarted the problem we can rectify
i am using the System image file is "flash:c3745-ipvoice-mz.124-1.1.bin"
any updates of flash can reolve this issue .

Configuration for voice depends on the router platform.Make sure the configuration is fine.Refer URL
http://cisco.com/en/US/products/ps6441/products_configuration_guide_chapter09186a0080541bc6.html#wp1008931

Similar Messages

What is the command to check if there are any active calls before restarting the voice router?

what is the command to check if there are any active calls before restarting the voice router? thanks

Hi.
I can suggest show call active voice or show voice call status or show sip-ua call brief in case of SIP TSP.
HTH
Regards.
Carlo

Error in voice router

voice router 3745
flash:c3745-ipvoice-mz.124-1.1.bin
i am getting the error and router restarted automatically
Apr 15 02:52:22: %DSPRM-5-SETCODEC: Configured codec 8 is not supported with this dsp image
kindly help to sort out this issue

Hi,
can you upgrade to latest 12.4 and try again ?

Voice Routing: Normalization Rule vs. Route

We're just testing some Enterprise Voice stuff within our Lync Environment but there's still one Thing not clear to me: On the specific user tab, we're able to assign a "Dial plan policy" and a "Voice policy"
In the Dial plan policy we're able to enter the normalization rules for Digit Manipulation.
In the Voice policy we can assign routes the call will take (e.g trunk).
But for example if there is a normalization rule which translates +402221111 to 1111 and on the route there are just numbers allowed starting with 1, why the call will fail? In my opinion the number should be translated to 1111 right after dialing and
this should match the proper route.
Can someone describe how the callflow is working in Detail. Will it maybe check for the specific route before there is any Digit Manipulation process?
Thanks in advance

Typically you'd do it the other way around, you'd translate 1111 to +402221111 so that it's in proper E.164 format, and you'd have your routes match that.
What you described should work though, at a very high non-mechanical level:
A number is typed into Lync
The number is normalized by the users dial plan (typically to a standard such as E.165)
The normalized number is compared against the voice policy, which will match if it finds a route that matches that's tied to the policy through a PSTN usage.
An INVITE is sent through the trunk using the path determined.
What kind of errors are you getting and do you get a "pass" in the Lync Control Panel -> Voice Routing -> Test Voice Routing when you enter the information? If not, you've got a typo or misconfiguration somewhere and giving more detail
will help.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

TFTP error on 3925 intergrated voice router

We have a 3925 intergrated voice router that prompts this error cmapp_sccp_tftp_download_file: Unable to get configuration file size tftp://x.x.x.x/SKIGxxxxxx.cnf.xml, rc=-2
It gives the IP to our call manager then a mac address of analog gatway and extension pertaining to cnf.xml. The error runs when the router is connected to the network and also when it is not connected to the network. I have looked in the Cisco bug toolkit did not find a bug that was related to the issue. I check the tftp in the call manager there is no file of that type which there should not be. We have 3925s voice routers already registered to the call manager. Voice router information below:
Cisco IOS Software, C3900e Software (C3900e-UNIVERSALK9-M), Version 15.2(4)M2, RELEASE SOFTWARE (fc2)
System image file is "flash:c3900e-universalk9-mz.SPA.152-4.M2.bin"
Cisco CISCO3925-CHASSIS (revision 1.0) with C3900-SPE200/K9 with 1797120K/300032K bytes of memory

No, we are not using as CME.
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2015.03.26 11:13:02 =~=~=~=~=~=~=~=~=~=~=~=
XXXXXXXX#
XXXXXX#term len 0
XXXXXXXX#sh run
Building configuration...
Current configuration : 14446 bytes
! Last configuration change
! NVRAM
! NVRAM
version 15.2
no service pad
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
service password-encryption
service sequence-numbers
hostname XXXXXXX
boot-start-marker
boot system flash:c3900e-universalk9-mz.SPA.152-4.M2.bin
boot-end-marker
card type t1 0 0
logging buffered 16384
enable secret 4 DJsu7i99xp.Xzvkf3epKdaTCS5CUguxaCh2jhXROMZ.
aaa new-model
aaa authentication login default group tacacs+ local
aaa authentication enable default group tacacs+ enable
aaa authorization console
aaa authorization exec default group tacacs+ if-authenticated
aaa authorization commands 15 default group tacacs+ none
aaa accounting exec default start-stop group tacacs+
aaa accounting commands 15 default start-stop group tacacs+
aaa accounting connection default start-stop group tacacs+
aaa accounting system default start-stop group tacacs+
aaa session-id common
clock timezone CST -6 0
clock summer-time CDT recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
ip cef
no ip domain lookup
ip domain name textron.com
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-dms100
voice-card 0
dspfarm
dsp services dspfarm
voice rtp send-recv
voice service pots
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g729br8
voice translation-rule 1
rule 1 /3451/ /918005336093/
voice translation-rule 2
rule 1 /^$4...$$/ /XXXXXX\1/
rule 2 /^$6...$$/ /XXXXXX\1/
voice translation-profile 5-digit-dial
translate called 1
voice translation-profile pstn-in
translate called 2
application
global
service alternate default
license udi pid
hw-module pvdm 0/0
hw-module pvdm 0/1
hw-module pvdm 0/2
vtp domain BHAMA
vtp mode transparent
redundancy
controller T1 0/0/0
cablelength long 0db
pri-group timeslots 1-24 service mgcp
description
controller T1 0/0/1
cablelength long 0db
pri-group timeslots 1-24 service mgcp
description
csdb tcp synwait-time 30
csdb tcp idle-time 3600
csdb tcp finwait-time 5
csdb tcp reassembly max-memory 1024
csdb tcp reassembly max-queue-length 16
csdb udp idle-time 30
csdb icmp idle-time 10
csdb session max-session 65535
interface Loopback0
ip address
interface Loopback11
description
ip address
interface GigabitEthernet0/0
description
ip address
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1
description
ip address 1
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
interface GigabitEthernet0/3
no ip address
shutdown
duplex auto
speed auto
interface Serial0/0/0:23
description
no ip address
encapsulation hdlc
isdn switch-type primary-dms100
isdn incoming-voice voice
isdn map address .* plan isdn type national
no cdp enable
interface Serial0/0/1:23
description
no ip address
encapsulation hdlc
isdn switch-type primary-dms100
isdn incoming-voice voice
isdn map address .* plan isdn type national
no cdp enable
router eigrp 1
network 10.0.0.0
redistribute connected
passive-interface default
no passive-interface GigabitEthernet0/0
no passive-interface GigabitEthernet0/1
ip forward-protocol nd
no ip http server
no ip http secure-server
ip tacacs source-interface Loopback11
logging trap notifications
logging facility local1
logging source-interface Loopback11
logging host
logging host
logging host
logging host
logging host
access-list 18 permit
access-list 18 permit
access-list 18 permit
access-list 18 permit
access-list 86 permit
access-list 86 permit
access-list 86 permit
access-list 86 permit
access-list 89 permit
access-list 89 permit
access-list 89 permit
nls resp-timeout 1
cpd cr-id 1
snmp-server community
snmp-server community
snmp-server ifindex persist
snmp-server trap-source Loopback11
snmp-server locatiON
snmp-server contact
snmp-server system-shutdown
snmp-server enable traps snmp authentication linkdown linkup coldstart warmstart
snmp-server enable traps vrrp
snmp-server enable traps flowmon
snmp-server enable traps transceiver all
snmp-server enable traps ds1
snmp-server enable traps call-home message-send-fail server-fail
snmp-server enable traps tty
snmp-server enable traps eigrp
snmp-server enable traps gatekeeper
snmp-server enable traps xgcp
snmp-server enable traps license
snmp-server enable traps ethernet cfm cc mep-up mep-down cross-connect loop config
snmp-server enable traps ethernet cfm crosscheck mep-missing mep-unknown service-up
snmp-server enable traps flash insertion removal
snmp-server enable traps auth-framework sec-violation
snmp-server enable traps ds3
snmp-server enable traps entity-sensor threshold
snmp-server enable traps isdn call-information
snmp-server enable traps isdn layer2
snmp-server enable traps isdn chan-not-avail
snmp-server enable traps isdn ietf
snmp-server enable traps adslline
snmp-server enable traps vdsl2line
snmp-server enable traps c3g
snmp-server enable traps icsudsu
snmp-server enable traps envmon
snmp-server enable traps ds0-busyout
snmp-server enable traps ds1-loopback
snmp-server enable traps energywise
snmp-server enable traps vstack
snmp-server enable traps mac-notification
snmp-server enable traps aaa_server
snmp-server enable traps atm subif
snmp-server enable traps cef resource-failure peer-state-change peer-fib-state-change inconsistency
snmp-server enable traps memory bufferpeak
snmp-server enable traps cnpd
snmp-server enable traps config-copy
snmp-server enable traps config
snmp-server enable traps config-ctid
snmp-server enable traps entity
snmp-server enable traps fru-ctrl
snmp-server enable traps resource-policy
snmp-server enable traps event-manager
snmp-server enable traps frame-relay multilink bundle-mismatch
snmp-server enable traps frame-relay
snmp-server enable traps frame-relay subif
snmp-server enable traps hsrp
snmp-server enable traps ipmulticast
snmp-server enable traps msdp
snmp-server enable traps mvpn
snmp-server enable traps nhrp nhs
snmp-server enable traps nhrp nhc
snmp-server enable traps nhrp nhp
snmp-server enable traps nhrp quota-exceeded
snmp-server enable traps pim neighbor-change rp-mapping-change invalid-pim-message
snmp-server enable traps pppoe
snmp-server enable traps cpu threshold
snmp-server enable traps rsvp
snmp-server enable traps syslog
snmp-server enable traps l2tun session
snmp-server enable traps l2tun pseudowire status
snmp-server enable traps vtp
snmp-server enable traps bstun
snmp-server enable traps dlsw
snmp-server enable traps ipsla
snmp-server enable traps stun
snmp-server enable traps bfd
snmp-server enable traps bulkstat collection transfer
snmp-server enable traps mpls traffic-eng
snmp-server enable traps mpls fast-reroute protected
snmp-server enable traps mpls rfc ldp
snmp-server enable traps mpls ldp
snmp-server enable traps pw vc
snmp-server enable traps ipmobile
snmp-server enable traps snasw alert isr topology cp-cp port link dlus
snmp-server enable traps dial
snmp-server enable traps dsp card-status
snmp-server enable traps dsp oper-state
snmp-server enable traps dsp video-usage
snmp-server enable traps dsp video-out-of-resource
snmp-server enable traps ethernet cfm alarm
snmp-server enable traps rf
snmp-server enable traps vrfmib vrf-up vrf-down vnet-trunk-up vnet-trunk-down
snmp-server enable traps mpls vpn
snmp-server enable traps ccme
snmp-server enable traps srst
snmp-server enable traps voice
snmp-server enable traps dnis
snmp-server host
snmp-server host
snmp-server host
tacacs-server host
tacacs-server host
tacacs-server host
tacacs-server host
tacacs-server key
control-plane
voice-port 0/0/0:23
voice-port 0/0/1:23
ccm-manager fallback-mgcp
ccm-manager redundant-host 10.244.200.33 10.227.210.33
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold bind Loopback11
ccm-manager config server X.X.X.X. X.X.X.X X.X.X.X
ccm-manager config
ccm-manager sccp local Loopback11
mgcp
mgcp call-agent X.X.X.X. 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability mf-package
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
mgcp default-package line-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp fax t38 nsf 000000
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface Loopback11
mgcp bind media source-interface Loopback11
mgcp profile default
sccp local Loopback11
sccp ccm X.X.X.X identifier 1 priority 1 version 7.0
sccp ccm X.X.X.X identifier 2 priority 2 version 7.0
sccp ccm X.X.X.X identifier 3 priority 3 version 7.0
sccp ip precedence 3
sccp ccm group 1
bind interface Loopback11
associate ccm 1 priority 1
associate ccm 2 priority 2
associate ccm 3 priority 3
associate profile 2 register Amarillo_XCODE
associate profile 3 register Amarillo_MTP
associate profile 1 register Amarillo_CNFB
switchback method graceful
switchback interval 8
dspfarm profile 1 transcode
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 20
associate application SCCP
dspfarm profile 2 conference
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 8
associate application SCCP
dspfarm profile 3 mtp
codec g711ulaw
codec pass-through
maximum sessions hardware 42
maximum sessions software 200
associate application SCCP
dial-peer voice 1 pots
translation-profile incoming pstn-in
incoming called-number .
direct-inward-dial
dial-peer voice 10 pots
destination-pattern 9[2-9]11
port 0/0/0:23
forward-digits 3
dial-peer voice 11 pots
preference 1
destination-pattern 9[2-9]11
port 0/0/1:23
forward-digits 3
dial-peer voice 12 pots
destination-pattern 911
port 0/0/0:23
forward-digits 3
dial-peer voice 13 pots
preference 1
destination-pattern 911
port 0/0/1:23
forward-digits 3
dial-peer voice 20 pots
destination-pattern 9[2-9].........
port 0/0/0:23
forward-digits 10
dial-peer voice 21 pots
preference 1
destination-pattern 9[2-9].........
port 0/0/1:23
forward-digits 10
dial-peer voice 30 pots
destination-pattern 91[2-9]..[2-9]......
port 0/0/0:23
forward-digits 11
dial-peer voice 31 pots
preference 1
destination-pattern 91[2-9]..[2-9]......
port 0/0/1:23
forward-digits 11
dial-peer voice 40 pots
destination-pattern 9011T
port 0/0/0:23
prefix 011
dial-peer voice 41 pots
preference 1
destination-pattern 9011T
port 0/0/1:23
prefix 011
dial-peer voice 50 pots
incoming called-number .
direct-inward-dial
gatekeeper
shutdown
call-manager-fallback
secondary-dialtone 9
max-conferences 8 gain -6
transfer-system full-consult
ip source-address X.X.X.X port 2000
max-ephones 700
max-dn 1400 octo-line
system message primary Operating in SRST mode
system message secondary In SRST mode
transfer-pattern .T
voicemail XXXXXXXXXXXX
translation-profile incoming 5-digit-dial
call-forward pattern .T
call-forward busy XXXXXXX
call-forward noan XXXXXXXX timeout 10
time-zone 8
banner exec ^C
^C
banner login ^C
******************* WARNING NOTICE *******************
^C
line con 0
exec-timeout 15 0
password
line aux 0
exec-timeout 15 0
password
line vty 0 4
exec-timeout 60 0
password
transport input all
line vty 5 924
exec-timeout 60 0
password
transport input all
scheduler allocate 20000 1000
ntp access-group
ntp server
ntp server
ntp server
ntp server

Voice router reboot automatically

Hi all,
I have cisco router C1841 IOS 12.4, when I made a voice call through the router it reboot automatically, but download and internet conection work fine. I already replaced three times (router, modem) but the problem still occur. is every one can help me?
Thanks in advance,
Regards,
Rja

Hi Paolo,
I don't have a service contract for C1841 to download the update, Give me please if you are able to download it.
IOS c1841-advsecurityk9-mz.124-22.YB5.bin is running in my router now.
Thanks,
Rja

SPA 3102 Voice router and gateway

Hello, I was told to get this device but i do not know what is does and I did not find information on the Linksys website about its purpose. I am setting up a desktop PC as a webserver thru a cable conection that does not have a static IP. My understanding is that this device will cause it to behave like a static IP. Is this correct?
I followed the instructions to connected it but I don't know how to fine the "DNS" numbers to imput into the field
I am running Windows Vista Home edition as the OS. The server will run a voice response system.
Any suggestions.

First of all, you can get the information for the device on this site.
http://www.cisco.com/en/US/products/ps10027/
This unit is part of Cisco-Linksys Small Business Voice Gateways and ATAS.
By the way, static IP has nothing to do with the SPA3102. If ever you'd like to setup a
webserver, it requires you to set a static IP on the server itself (IP of the PC) and at least a
Static IP for your WAN connection. WAN Static IP will be provided by your ISP and you need
to subscribe that IP from them.If in case your server requires you to open a port number for
the Server then that's the time you forward it to the router.

DRAM slot usage on 1760 Voice router

I have a 1760 router with 96MB DRAM. Is there a way I can find out, without opening the router up, (1) how many slots there are for installing DRAM; and (2) which of the slots are occupied with DRAM and which are free?

I think with 1760, only one DIMM slot is available and in ur router, it is fixed with 64 MB RAM.This router already has 32MB on board by default; plus the 64Mb DIMM; total 96Mb.SO, to upgrade u may have to remove that 64MB and replae it with the memory size u want. But as of now, there is not any command that will tell you which banks or slota are filled for dram. They are looking into implamenting this in future ios's. But as of now there is no command.

Pbx to voice router connection...?

Hi
i`m facing some problem in the detail connction between a pbx system and voice gateway to make calls from pbx system go via wan instead of the
pstn.
what module will be installed in both, the pbx and the voice gateway and how to configure both of them.
Fxs port on pbx will be connected to fxo port on the gateway....?
is a vwic module will be installed to the gateway..? what type..? what is multiflex and drop and insert..?
Can you guide please to a link (pdf) to know all of these details...........?
thanks for your permanent help.

thanks for replay,
what i need is only some help ..
i`m a cisco voice certified and i don`t have sufficient experience as i`m only installing my first project.
i studied very hard and i`m having a lot of concept and info but you konw the feeling when you install and configure
a physical devices for the first time and you may forget simple things and concepts.
hope u got my point and thanks again.

Voice Routing and Normalization Rules

Hello All,
I am very weak about Lync ENT voice. I want to configure the Lync 2013 for below scenarios.
PABX extensions – 2xx, 3xx.
Lync extensions – 4xx
Access code – 9 for external access (This will be added from the AudioCodes gateway)
What should i configure ?

please may i know the Build Numbers of latest patches ?
i have Cu4 on of Lync 2013 Server.
Should i want to add below line ?
$x = New-CsClientPolicyEntry -Name "ShowExtensionInFormattedDisplayString" -Value "True"
$y = Get-CsClientPolicy -Identity Global
$y.PolicyEntry.Add($x)
Set-CsClientPolicy -Instance $y

Lync Enterprise with Single BE Server and Voice Routing

Hi team,
I read that the best way for a HA topology is to go ahead with 3 FE server pool(Although MS has said it's workable, many recommend not to). Now this leaves me with another problem, can I go ahead with just 1 SQL Back-End server. I know in this case SQL will
have no failover. But if we ok with the downtime, will it work properly when the BE is up?
The front-end includes the following:
Basic Lync functions
Mediation
Monitoring
Archiving
Thank you.
Chris!

Hi Crypto_J,
After the Back End Server is up, you could verify if the services of Lync Servers are normally started, and take a test.
For more details about restoring the Back End Server, please click on the link below.
Restoring the server hosting the Central Management store in Lync Server 2013
http://technet.microsoft.com/en-us/library/hh202172.aspx
To ensure high availability for your Back End Servers, you can use either synchronous SQL mirroring or SQL clustering.
Using one of these solutions optional, but is recommended to maintain your organization's business continuity.
Best regards,
Eric

Cisco Voice enabled Router

Hello,
I would like to know how to configure a voice router in terms of translation rule and then option 150 .
Please explain in details.
Thanks in advance.

Voice translation-rule/profile:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
Option 150:
http://www.cisco.com/en/US/docs/ios/12_2/configfun/configuration/guide/fcf002.html#wp1020232
After reviewing these documents, if you have any questions, ask away.
HTH
Adam
remember: please rate if helpful

New Dial plan & Voice policies not taking effect with Polycom CX 600 Desktop Phone in production deployment, Worked fine in Testing

Hi,
We are in the process of Migrating Cisco CUCM & Voice Gateway (From another vendor to Cisco).
The requirement is all internal calls between Cisco IP Phones & Lync to be flown through CUCM. Means internal extension to extension. Remaining all calls like Mobile, National, International, Toll Free, Emergency, Shared numbers calling to be routed
to Cisco Voice Gateway.
We created the test dial plan, Voice policies, Route and assigned it to couple of user from Lync (2 extensions) and from Cisco side we have taken 2 IP Phones which is pointed to new CUCM. We tested all below scenarios,everything was working fine.
Lync to Lync Call using internal Extension number – Routed through Cisco new CUCM
Lync to Cisco Call using internal Extension number – Routed through Cisco new CUCM
Cisco to Lync Call using internal Extension number – Routed through Cisco new CUCM
Lync to Hotline Numbers (66XX, 68XX Numbers) – Routed through Cisco Gateway
Lync to Shared Numbers starting with 600 (Verified the number 600535353) - Routed through Cisco Gateway
Lync to Emergency numbers & Toll Free Numbers (Not verified the emergency Number as we decided to do it at end) - Routed through Cisco Gateway
Lync to Landline Numbers – Any 7 digit numbers - Routed through Cisco Gateway
Lync to National Numbers – Starting with 3,4,6,7,8 followed by 7 digits - Routed through Cisco Gateway
Lync to Mobile Phones – Starting with 05 contains exactly 10 digits - Routed through Cisco Gateway
Lync to International Numbers – Starting 00 contains at least 11 digits - Routed through Cisco Gateway
All Incoming calls – From Landline, Mobiles, International Numbers - Routed through Cisco Gateway
Call Transfer – To another Lync Extension, Cisco Extension, Landline, Mobiles, International Number
Conference – with another Lync Extension, Cisco Extension, Landline, Mobiles, International Number
Call Forwarding – To another Number, Voice mail
Response Groups
Click to call – As if user try to place a call by directly click the number from Outlook, Websites will be in E.164 format
Dial in meeting – Conference calls are works fine
But when we roll out to the production we are facing issues listed below
1) The phones we used during testing are working which is using same dial plan, Voice policy, Route, PSTN Usage. But from production most of the phones are not working (using the same dial plan, voice policy, Route). Also Problem is only with external calls
as the internal calls are working fine between Cisco & Lync even in production (Routed through CUCM) NOTE: All incoming calls are working fine (From international, local, national, extension)
2) How long its going to take for Lync to push the new voice policies, Dial plans to the Phones?
3) Is there a way to forcefully update the policies, dial plans to the Phone?
4) Also the environment is using over 100 dial plans, so I just copied and pasted the Normalization rules that we tested and working fine. Most of the dial plans are assigned to individual users as every dial plan contains a normalization rule for
international calling with Unique Prefix (Example: User John international Normalization rules says #1234#00#CountrycodePhonenumber, means if John has to place the international call he need to dial #1234# followed by 00 and then country code, then actual
phone number). In this case how long its take for the users / phones to get updated with new dial plans?
6) Is it recommended to use multiple dial plans ? What are the best practices?
5) Also calls are working fine one & failing on subsequent tries. Means when I dial first 1 or 2 times. Call fails, but when I try 3rd time and subsequently it works. After some again there will be failure during 1 or 2 attempts. Why is it so?
6) After updating the dial policies, voice Route, Voice policies If i reboot all the phones from Switch, Will the changes take effect immediately?
7) Also when some one calling from mobile or external number to Lync extensions they cant here any Dial tones or caller tunes? Its working fine when they call Cisco Extensions. Also to Lync its working if we dial in E.164 Format, if we dial like 023XXXXX
format its not working. Any guess about this issue?
Waiting for some one to help,
Best regards
Krishna
Thanks & Regards Krishnakumar B

Hi,
1. As all incoming call worked normally, please double check outgoing ports for Lync FE Server and Mediation Server.
You can refer to the link of “Ports and protocols for internal servers in Lync Server 2013” below:
http://technet.microsoft.com/en-us/library/gg398833.aspx
2. When an administrator makes a change to Lync Server (for example, when an administrator creates a new voice policy or changes the Address Book server configuration settings) that change is recorded in the Central Management store.
In turn, the change must then be replicated to all the computers running Lync Server services or server roles.
So it may not replication completely immediately.
3. You can run the following cmdlet with Lync Server Management Shell on FE server to
forcibly replicate information to a computer: Invoke-CsManagementStoreReplication
4. As you used over 100 dial plans, it may be the issue of multiple dial plans. Would you please tell us why you created different dial plan for individual user with unique prefix?
5. Multiple dial plans and undue normalization rules may cause call fail. You can double check the normalization rule.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

How does a single cisco router network enable outgoing calls for small businesses if no dial plan is used

Hi Cisco techs,
I was recently asked this question at work from the rest of my team as they know I am doing studies for cisco ICND 1 & 2.
And couldn't answer them.
Or does this require more information like hardware used etc?
Had searched Google to ambiguous results.
Can someone give me a link to advise please
Thanks in advance again..

Dear,
For small business (up to max 450 users) we use Call Manager Express (CME) which is configured on Cisco Voice router.
If no dial plan is used or configured then none of callmanager can do external outgoing/internal calls. Dial plan is basic thing which must be configured on cisco voice router (e.g. CME).
If you have configured Phones (ephone & ephone dn) and if they are registered on callmanager express then for internal calls you dont need to configure dial plan because callmnager have all the information of its phones.
If you want to have external incoming/outgoing calls for CME then you will have to create dial peers/dial plan.
For a CME lab setup, you can visit below link.
http://cisco.jjc.edu/cnt208/PDF/CCNP4_lab_2_1_en.pdf
Suresh

Changing CAS e&m-wink-start to a PRI on voice gateway

Cisco IOS Software, C3900 Software (C3900-UNIVERSALK9-M), Version 15.1(4)M4, RELEASE SOFTWARE (fc1)
I want to change a CAS e&m-wink-start to a PRI on controller T1 0/1/0 using mgcp below. I am posting the new configuration below the current e&m-wink-start configuration on the voice gateway router.
Current
controller T1 0/1/0
cablelength long 0db
ds0-group 1 timeslots 1-24 type e&m-wink-start
description
New configuration
Router(config)# no contoller T1 0/1/0
Router(config)# no interface Serial0/1/0
Router(config)#controller t1 0/1/0
Router(config-controller)#cablelength long 0db
Router(config-controller)#framing esf
Router(config-controller)#linecode b8zs
Router(config-controller)#clock source line
Router(config-controller)#pri-group timeslots 1-24 service mgcp
Router(config-controller)#description circuit ID
Router(config-if)# interface serial0/1/0
Router(config-if)# no ip address
Router(config-if)# encapsulation hdlc
Router(config-if)# isdn switch-type primary-4ess
Router(config-if)# isdn incoming-voice voice
Router(config-if)# isdn bind-l3 ccm-manager

Hi,
You need additional configs for this to work..You also need to configure the isdn bind on the D-channel not as you have it..
Router(config-if)# interface serial0/1/0:23
Router(config-if)# no ip address
Router(config-if)# encapsulation hdlc
Router(config-if)# isdn switch-type primary-4ess
Router(config-if)# isdn incoming-voice voice
Router(config-if)# isdn bind-l3 ccm-manager
Additional configs....change ip to suit your needs
ccm-manager redundant-host 192.168.103.114
ccm-manager mgcp
ccm-manager music-on-hold bind xxx--put relevant interface
mgcp
mgcp call-agent 192.168.103.115 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp modem passthrough voip codec g711alaw
mgcp modem passthrough voip redundancy
mgcp ip qos dscp af31 media
mgcp ip qos dscp cs3 signaling
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
mgcp default-package dtmf-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
mgcp tse payload 100
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interfacexx -----------------------put relevant interfcae here
mgcp bind media source-interface xxx--------------------same
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"

Voice router

Similar Messages

Maybe you are looking for