- While copying and pasting the DHCP configs, forgot to change the default gateway for second scope
- Did not set ntp master in router
- forgot to set ntp in subscriber, had to go back and do it again
- forgot to set framing and linecoding in controller
- did not shut the voice port before removing ds0-group
- forgot call pickup group and night service
- ip phones with night service, not clear??? Is it for ephones or ephone-dns
- when integrating unity with CME or CCM, always verify servers before saving the configs, because when you save unity services will restart
- forgot to reset voicemail hunt list after adding,
- forgot to change the framing at PSTN switch
- at CME, the call was matching the wrong dial-peer.
- forgot to set portfast for CCM and unity ports
- forgot to add DNS entry in unity
- forgot to add ds0 country code
- forgot to add 91 to the after hours block pattern in CME
- forgot to enter secondary dialtone
- configured night service to 8:00 instead of 7:59
- use bulk import wizard to add subscribers
- infrasturcture – 18 minutes 3 sites, setting vlans, assigning to ports, trunking etc, dhcp pools.
- messed up ip addressing in branch 2.
- dont set mwi on-off first before setting number command in ephone-dn
- created so many errors in bulk subscriber import
- alias,lastname,firstname,dtmf
- created live record first…but no way to send message to a PDL.
- So created a call handler with extn 1000 and then set the greeting to blank, after message, toold it to take a mesage and message receipient was set to all subscribers PDL. After message, send caller to hangup.
- mwi is not working for callmanager subscribers. restarted unity to fix this.
- route-group route list name incompatibility. deleted route groups and added it back fixed it.
- use no ip domain-lookup on all routers.
- for e1 r2, the first dial peer was alwasy gtetting matched (which had a destination pattern configured). And ANi was set to the destination pattern number in the first dial -peer. To avoid this situation, alwasy configure first dial peer with incoming-called number . so that this problem can be avoided.
- used full international digits to dial branch 2.instead use 4 digit dialing. and prefix the 011331322 to the dialed digits (3xxx).
- after hours block pattern i used 1900, 1976. instead should use 91900,91976 (because 9 is dialed).
- night service – specify night-service bell on the source ephone-dn
- specify night-service bell on the destination ephones.
Archive for the ‘Lab sessions’ Category
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- How to configure ATA for voice VLAN?
- Set speed and duplex for all ports
- Do we need to configure spanning tree port fast or not for ip phone ports on NM-16ESW? Yes, configure spanning tree for NM-16
- To force IP Phones to renew DHCP, disable and enable inline power
- Overlooked the fact the BR2 is CME and configured dhcp option 150 to CCM
- Always configure publisher as the auto registration server. Did not work for Subscriber
- MSFC retains the access list applied to interface vlan even after the vlan interface was removed
- Did not consider voice mail ports for max ephone and dn when doing the CME setup
- Forgot to set the Callforward settings on phone, during initial setup
- did not notice crs admin user name, created the usual user rather than what was in the question
- no need to install crs editor, it is already installed
- ICD extensions and shared lines??
- configured the wrong agent user
- forgot about ip phone agent telecaster user
- CRS answers calls, calls agent, but does not patch call – Issues with jtapi client, had to update it using the jtapi client update tool
- Idea – for easy editing of phone line settings create a dummy phone and add all line to that phone. You need to select only one phone and toggle between lines
- Forgot to add EM Service before creating EM Profile, had to come back and do it.
- When copying the EM service link from univercd, copy the example link since it has the entire link in one line
- Configured AC user with password as cisco
- Did not associate the attendant phone to ac user
- Forgot to check call park retrieval for ac user
- Change DNS to IP in update plugins
- attendant console, calls to pilot number should be transferred to voicemail on no-answer. I configured the last member as voicemail pilot. This is wrong. You need to configure the call forward no answer on the hunt group members to voicemail and create a subscriber in unity with the extension of hunt pilot. Set the TCD Service parameter, Reset original called number on redirect to False
- clearing a vlan doesnt remove the actual config from the port in 6500
- set ata port also to trunking (vlan, auxiliary vlan).
- didnt put spanning-tree portfast on all switchports (nm16esw and 3550sw)
- lease 0 8 — means 8 hours, lease 8 means 8 days.
- dont use auto registration – time waste
- wasted 1.30 hours registering phones
- configured wrong helper address at hq
- unity message recording – set this in subscriber template (under each subscriber also) and under default subscriber COS.
- used label command instead of description in cme… should have used description command.
- modified default icd script by accident– dont do this…save as a new file and modify.
- applied wrong file in the application..
- put mgcp fax rate 7200 — but this is for t38 only.
- didnt use telephony-service setup command to setup CME. Should use this to reduce time.
- check against trivial passwords for security – this should be unchcekd so that you can set password to 54321.
- call restriction table – ???? or 4 digits as maximum transfer digits.
- Subscribers -> Account policy.
- attendant console – created a 3rd member, for voicemail transfer. No need for this. when somebody doesnt answer the call it will roll over to voicemail and hit the pilot’s mailbox.
- did uncheck fax relay ECM override on 6608. TO disable ECM this should be checked
- didnt set fax relay speed settings to 7200 in vg248. this should be set to 7200bps. (global setting).
- if far end gateway is 6608 set nsf to preserve in vg248
- interface vlan — give no shut
- line vty >> exec-timeout 0 0
- check in phones if DHCP is enabled
- give ip helper-address. If you enter it wrong, take it out using no command before entering the new one, because it does not overwrite
- IPPhone auto registration. Auto registration was enabled in pub, but in the Callmanager group, pub was listed second. Hence phones did not register. Always use Pub for auto registration and list Pub first. After the phones register, disable autoregistration and rearrange the callmanager group.
- Autoregistration, use dummy phone range so that you will not run into shared line issues later. Turn off auto registration after the phones get registered
- Did not configure IOS DHCP along with DHCP Server
- ATA was not DHCP enabled
- enter isdn switch-type before moving to controller mode
- linecode and framing defaults are sf and ami, need to change this
- Use codec preference in dial-peers so that calls from the gateway to phones at the same site can use g711 and calls to remote sites can use g729
- Check CAC, codec and SRST section and add regions and locations and then create the device pool and add all these to the phones and gateways at the same time. Remember to set Call forward on lines to VM
- Add external phone number mask to phones
- Did not configure voice class for h323 dial-peer redundancy
- Did not apply CSS to phones
- When making changes to phones, do not reset them individually. Make changes to all phones and then reset them all at the same time.
- Check Urgent Priority for 911 and 9911
- Shared lines does not sync external phone number mask. Check the Update Shared device settings
- Forgot to add pots dial-peers on BR2 gateway.
- Forgot to add TEHO partiton to BR2-Ph2.
- Enable dependency records in Enterprise parameters. Change all DNS names to IP Address
- AAR does not need any additional partitions or CSS. You can apply the least restrictive CSS to the AAR CSS as well as either the Long distance or international partitions depending on the scenario.
- Do all calling party and called party transformations at Route list
- Forgot to add Prefix digits to HQ route group in BR2-local route list
- TEHO Route pattern should have been 90111212.[2-9]XXXXXX, added it as 90111212.XXXXXXX
- If Telco is sending 10 digits, set the Significant digits in gateway to 4 (to match internal extension length) or create individual translation patterns to convert 10 digits to 4 digits.
- Forgot to enable Display IE delivery on H323 gateway
- Forgot to enable default application in H323 gateway for MGCP fallback
- Forgot mgcp dtmf-relay voip codec all mode out-of-band
- Forgot to add voice port to service mgcp dial-peer
- Forgot to add dual-line in max-dn srst
- Forgot voice rtp send-recv
- set port auxiliary vlan should be used on phone ports
- set route pattern from branch 2 to site 1 teho as long distance. But it should have been setup as international call.
- do not need direct inward dial on all dial peers, use this with incoming called number in one dial-peer
- voice class h225 timeout was set to 10 change this to 3
- forward digits set it to 7 for 7 digit local dialing
- on h323 gateways add dial peer for 911 separately. otherwise calls will fail if predot is set at route pattern level.
- set caling party extph mask setting on route list. as number of route lists will be less. number of route patterns will be too many so checking ext phone number mask at each route pattern will take more time.
- always put prefix digits in route list in called party
- forgot to check enable status poll on mgcp gateways (fractional)…
- didnt set aarcsss in gateways (br2 and 6608)
- set h323 gateway route patterns to discard digits NONE if gateway dail-peers are 9T
- set h323 gateway with dial-peers (individual dial peers) or .T as the destination pattern if using discard digits Predot at route pattern
- forgot 9911 in srst gateway
- used 2 extra dial peers for 4 digit dialing. instead can use num-exp
- used num-exp to map incoing calls (10 digit to 4 digits). dialplan pattern is enough…
- when calls come from pstn on a h323 gateway and it needs to send call to a local phone, the call needs to be g711. if it goes to a phone in another region, then that will need a g729 codec. use voice class codec to achieve this.
voice class codec 1
codec pref 1 g711ulaw
codec pref 2 g729r8
dial-peer voice 1 voip
destination-pattern 1…
voice-class codec 1