Debugging VOIP with Shikha Sharma #CiscoKid

Tell you what, this fixed my issue- Enabling "Check Progress Indicator Before Establishing Media" on call manager service parameter

We have been allocated three ranges of DIDs corresponding each PRI line say 0/0/0 for 1XXX, 0/1/0 for 2XXX and 0/0/1 for 3XXX The way I was using to show the correct DID in outbound calls is via local route group, of course I was using MGCP then. I recently moved from MGCP to H323 because of its cliched L3 issues causing rigorous call drop. After this change, I started facing the same issue with calling number again. Let me summarize here the fix I applied- it might look mussy! I created three partitions epitomizing my DID range say pt-1XXX pt-2XXX pt-3XXX Then I copied my route pattern for local/national/international and emergency numbers three times with these partitions like RP_9.T with pt-1XXX RP_9.T with pt-2XXX RP_9.T with pt-3XXX It will also require route list, corresponding each route pattern. In this route list, I defined calling number party mask as the full DID and discarding pre-dot digit replacing it with a unique digit. Like- RL_1XXX has calling part...

Jun 23 10:28:38 WST: ISDN Se0/0/1:15 Q931: TX -> DISCONNECT pd = 8  callref = 0x57CC     Cause i = 0x809B - Destination out of order This kind of error indicates partitioning issue on the number. If the number has no partition, provide it one gateway has access to. Otherwise, try resetting the phone. In my case, this error was occurring on only certain extensions without partition, weird but it restored.

The issue I am referring here is for incoming calls- PSTN line getting continuous tone Debugs will show that call gets connected And IP phone won't ring at all. ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref = 0x01DD         Sending Complete         Bearer Capability i = 0x8090A3                 Standard = CCITT                 Transfer Capability = Speech                 Transfer Mode = Circuit                 Transfer Rate = 64 kbit/s         Channel ID i = 0xA98388                 Exclusi...

In my scenario, it was only happening with physical phones and not with softphones. Hence, I took debug tftp events- .Mar  7 07:41:47.805: TFTP: Looking for CTLSEP001D70FC6A4E.tlv .Mar  7 07:41:47.885: TFTP: Looking for ITLSEP001D70FC6A4E.tlv .Mar  7 07:41:47.957: TFTP: Looking for ITLFile.tlv .Mar  7 07:41:48.093: TFTP: Looking for SEP001D70FC6A4E.cnf.xml .Mar  7 07:41:48.097: TFTP: Opened flash:/its/vrf1/ SEP001D70FC6A4E.cnf.xml, fd 4, size 1712 for process 343 .Mar  7 07:41:48.109: TFTP: Finished flash:/its/vrf1/ SEP001D70FC6A4E.cnf.xml, time 00:00:00 for process 343 .Mar  7 07:41:51.605: TFTP: Looking for Norwegian_Norway/be-sccp.jar .Mar  7 07:41:55.605: TFTP: Looking for Norwegian_Norway/be-sccp.jar .Mar  7 07:41:59.605: TFTP: Looking for Norwegian_Norway/be-sccp.jar .Mar  7 07:42:03.605: TFTP: Looking for Norwegian_Norway/be-sccp.jar .Mar  7 07:42:11.929: TFTP: Looking for Norway/g3-tones.xml .Mar  7 07:42:14.537: %IPPH...

Its not as difficult as it sounds. Its pretty simple actually. When you install Cisco Unity, it installs Windows exchange server too. Unity only supports Exchange or Domino as the message store options, all the voicemails run on exchange server. Also, webmail is enabled by default. All you need to do is following- 1. Get into Cisco Unity via mstsc 2. The URL to get into user's voicemail is- http:// <exchange IP>/exchange Exchange IP should be same IP as that of your unity. 3. Login with the user's credentials- Alias as his username and password. 4. If you don't know his password, reset it from AD users and computers. A Tip- If it doesn't show you anything in the inbox and stuck on loading, update java on server!

Here is how I configured it in our scenario- 1. Create a new voicemail profile for the user. Choose voicemail pilot and provide the Voice Mail Box Mask same as the extension of the user say 4626. 2. Add a new CTI route point for this user with right CSS and provide the DN as *4626. You may use any other prefix than *, it is used to identify the number as unique. 3. Choose the new voicemail profile which we created in step1 for this CTI RP and CFA calls to voicemail. 4. Make sure to provide right partition and CSS to enable this DN to reach Unity. Now, when an internal user dials *4626, it will take the call to the 4626's voicemail.

Let's begin this with configuration on Cisco Unity for the integration- Launch Cisco Tools Depot >> Switch Integration Tools >> Telephony integration Manager, double click on it to launch TIM. Now, a new window will open where we can integrate CME with the unity. Hit Create Integration >> SCCP >> Provide an Integration Name >> Next >> Provide IP address of the CME >> Next >> Next >> Provide the numbers for MWI on and off >> Next >> Number of ports you want to dedicate for this CME>> Next >> Next >> Finish . This will require us to restart ' AVCsGateway' and ' AvCsMgr ' service in Unity. After which the integration will complete. When Unity will complete the integration, it will generate voicemail ports on the concerned CME automatically, just like how it does in CUCM. You can check them with "show ephone register" Below is the sample of Unity ports- ephone-29[28...

 In order to directly transfer the call to user's voicemail, following is the sample configuration where *2... is *<extension> and 5000 is call-handler in CUE.   ephone-dn  181  number *2...  description **direct transfer to voicemail**  call-forward all 5000 voice translation-rule 5000  rule 1 /^.*\(....\)/ /\1/ voice translation-profile vmdirect  translate redirect-called 5000 dial-peer voice 5000 voip  description ** cue voicemail pilot number **  translation-profile outgoing vmdirect  destination-pattern 5000  session protocol sipv2  session target ipv4:10.22.32.12  dtmf-relay sip-notify  codec g711ulaw  no vad