A Small Note: CFwdALL Does Not Work for the Secondary Line

Yes, it’s true, you can’t forward calls on secondary lines from the phones. But one workaround is there, forward calls from ‘ccmuser’ or ‘ccmadmin’ page. This article from Cisco is much helpful for the tasks to do it:


Another good learning! 🙂


Default Codec for Voip Dial-Peer: G729

I didn’t know that! The default codec used by the voip dial-peers is G.729, I got the proof when I was calling from a PSTN phone towards H.323 branch router over T1 and passing it to central CUCM over voip. And also this was the reason I was not getting MoH at PSTN because MoH stream was multicasting over G711 only! Though my H.323 gateway and the destination phone was located on the same region but the call was establishing over G729. Then I found the issue after seeing the result of the command ‘sh voice call summary’:

Branch#sh voice call summary
PORT           CODEC     VAD VTSP STATE            VPM STATE
============== ========= === ==================== ======================
0/0/0:23.1      g729r8 n  S_CONNECT             S_TSP_CONNECT’

I was so surprised!! It’s the root cause my MoH is not working!!! I just asked one of my friend (Olivier: http://ccievoice.ksiazek.be) about the issue, he smiled and told me it’s normal to use g729 by default for the dial-peers unless I change from the voip dial-peer towards CUCM. Hats of my friend! I just changed the codec to G711 and it’s was working like magic.

dial-peer voice 9 pots
translation-profile incoming PstnInStripTo4
incoming called-number .
dial-peer voice 90 voip
destination-pattern 3…
session target ipv4: ! CUCM IP
codec g711ulaw
no vad

After configuring that:

SB#sh voice call summary

PORT           CODEC     VAD VTSP STATE            VPM STATE

============== ========= === ==================== ======================

0/0/0:23.1      g711ulaw n  S_CONNECT             S_TSP_CONNECT

0/0/0:23.2      –          –  –

And my MoH issue is fixed! It was a good learning for me. 🙂

Difference Between Barge and cBarge: In a line

‘Barge’ uses the built-in Conference Bridge on the phone, so allows maximum three parties on the conference.

‘cBarge’ uses the hardware/software Conference Bridge associated with CUCM/CME, so allows configurable number of users on the bridge.

It’s so simple, right?

A small note on AAR

AAR (Automated Alternate Routing) only works on the cases when remote site phones still have SCCP connection with the CUCM at the central site, but the remote location do not have enough bandwidth to establish a call over the IP WAN. It can reroute the call using AAR (via PSTN or secondary IP path) ONLY for this particular case. I just learnt it today when I tried to reach my SRST site phones by AAR, but just failed. But I accepted it cordially. 🙂

Import CUCM users to Unity Connection and UCCX

To Import at Unity Connection: Specify the ‘Primary Extension’ on the user page. Also active AXL service from both CUCM and CUCON.

To Import at UCCX: Specify the ‘IPCC Extension’ on the user page.

For both the cases pre-integration with CUCM is a must.

Mobile Connect/Single Number Reach (SNR) on CUCM

I love this feature, if you don’t know what this is I’m giving a short description on it. If someone calls you on your deskphone and SNR is configured like your mobile phone is associated with your deskphone, your mobile phone will also be connected after configurable number of seconds. So when you are moving somewhere from your desk and, your call will never me missed 🙂

Now let’s talk about it’s configuration, it’s pretty easier to configure, follow me:

1. Create an end user and associate his deskphone at his profile:

2. Configure Remote Destination Profile (RDP):

3. Configure Remote Destination:

That’s all! Just create a route pattern to route your mobile call toward destination.

Now let’s have a try calling your deskphone, your mobile will also be connected after configured seconds (4 seconds by default), if you answer your call from the mobile your deskphone can sense it! wohoo!! your deskphone line will lid red as long as you are talking over mobile. 🙂 okey, now think you have reached near to your desk and now you want to keep talking using your deskphone, so hangup the call from the mobile it will get hold automatically and now resume it from your deskphone. It’s cool, right? I loved this feature so much! 🙂