This post is a follow up to my posts on the Panasonic KX-UT670. (Link)

Now that I got Exchange working on the phone I was able to import all my contacts.
The problem now is when trying to call almost any of my contact’s numbers the call fails.

After some research I discovered the following; my contacts have their numbers formatted in the following manner: (888)555-1212.
When the KX-UT670 sends the call to the PBX it includes all the charterers in the phone field.
So instead of 8885551212 it sends (888)555-1212. When the PBX receives the call it can’t find a route and the call fails.

Hopefully Panasonic will release an update that will correct this.

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One of the biggest shortcomings of the Panasonic KX-UT670 is that there is no way to automatically sync your contact list and calender. (See my previous post: Link to post)
It does support IMAP and POP email but my assumption is that most users will not be sending emails from their office phone since they are sitting in front of their computer.

We use Exchange in my office so my goal was to get my contacts and calender synced to the phone.

I found a program called TouchDown from NitroDesk which is an Exchange ActiveSync client for Android.
Being that there is no Android Marketplace on the UT670 I had to navigate to NitroDesk’s website and download the .apk from from there.
Once it was downloaded and installed setup was a snap. I entered some information about my Exchange account and about five minutes later my emails, contacts, and calendar were all synced to the phone.

TouchDown includes Widgets that can be placed on the home screen. I have mine setup to show my calender and unread emails.

In my opinion this makes the UT670 worth purchasing. Without it I would have to manually sync my contacts with the SD Card which is pointless.

TouchDown is a relatively expensive app at $19.99 but without it I would not recommend purchasing the UT670.


———– Update 01/26/2012————-
Panasonic has been in contact with me regarding these issues and it seems that many of them will be addressed in firmware updates. Please see the follow up posts on the KX-UT670 at Link

The Panasonic KX-UT670 is an Android based 7″ SIP Phone.
It has all the bells and whistles you would expect of an Android phone.
The screen is bright and nice, interface is quick and snappy, and Panasonic provides a great SDK.

Here are the shortcomings I have found:
1) No Exchange support
2) No Bluetooth. Even though their site says it supports it I can not find a setting for it
3) Firmware upgrade does not work from the web admin. (The option is there but it doesn’t work)
4) Handset volume does not go loud enough
5) Screen is not dual touch (no pinch zoom) and does not scroll smoothly with light swipes
6) No marketplace
7) Can not add “Flexible Button” (BLF, speed dial, etc.) properties from the web admin
8) Officially supports only about 5 of Panasonic’s most expensive cameras (see next)
9) I can not get even the officially supported camera to work. There is an error saying that it does not support the camera’s settings but does not indicate what they should be
10) When you pickup the handset the “Check Voice mail” buttons disappears. You need to press the button and then pickup then handset
11) The sound quality is not as good as the other Panasonic IP phones. I am not sure if the handset is bad or it’s the phone
12) When you press the voice mail button and pick up the phone the dial pad disappears I then have to press the dial pad button before I can enter my password
13) Can’t make any changes without an SD card, install programs, change ring tones, etc but it doesn’t come with one
14) The touchscreen is a little off, I need to press below the key for it to register the correct key

At first looks it’s a great phone but the short coming above really limit its use.
Until at least some of these are corrected I would say stay away.

One of my Adtran Total Access 904 units stopped passing voice traffic and started producing the following errors:
2012.01.10 14:22:18 SB.CALL 8 RTP resource unavailable or SDP negotiation failed. Call from (XXXXXXX) to (YYYYYYYY).

A “show version” revealed the problem. It is supposed to show information about the DSP but it did not.

A normal operating unit will show something similar to the following:

slot 0, DSP 1
DSP software version: G1.A4.00.31
DSP hardware version: Freescale MSC7116
Total channels: 24

If it does not it means that the DSP failed and the unit needs to be returned to Adtran for repair.

Digium AA50 (S844B) 4 FXO 4 FXS

Posted: January 16, 2012 in IP PBX
Tags: ,

Here is my experience with the Digium AA50 over the past year.

It’s an Asterisk based PBX based upon the Blackfin processor.

It has all the features of Asterisk Business Edition in a small simple to setup package.

Base support is good but can take a day or two for a response.

Now for the cons:

Once a week the flash memory becomes read only and no can leave messages. The system then reboots itself!

I contacted Digium and they said it’s a known issue that will be fixed in a later update.

Another downside is that even though there is a WAN port it does not have any protection against SIP registration attacks.
Most PBX’s this size (Positron, Grandstream, and Yeastar) provide some level of protection.
When the system get’s attacked it can freeze up and/or stop accepting calls.

Hopefully these bugs will be fixed in a later release.

Recently a customer who installed a new Asterisk based PBX had the following issue with their door phone.

They hooked up their existing analog door phone to the FXO port on the PBX and had it route to a ring group.
When someone rings the bell the system sees it as an incoming call and rings the phones in the ring group. The person that answers the phone can then “buzz” the person into the office using their keypad.

The problem was that it took about five seconds before the phones in the ring group would ring.

After some research I figured out that the issue was that the system was waiting for caller id which was not supplied by the door phone. After three to four seconds it timed out and continued.

The fix was very simple; in the zapta.conf set usecallerid=no and the delay is minimal.

Here is a great post by Mark Holloway that lists the most commonly used debug commands for the Adtran Total Access:

http://www.markholloway.com/blog/?p=315

Debug Output

Command Description
show debugging shows current enabled debug commands
debug sip stack messages shows sip debug information
debug isdn l2-formatted shows isdn debug information (easier use)
debug voice verbose shows all voice messaging
debug isdn verbose shows all isdn messaging
debug interface T1 0/4 rbs shows robbed bit signaling from CAS circuit
undebug all turns off all debugging

Utility Commands

Command Description
clear count media counters on media-gateway reset
clear counters clears counters on interfaces
no events turns off messages to the current session
ping ping an IP address
traceroute trace the router path to a destination IP address
wall sends a broadcast message to all current users
sip trunk-registration force-register forces sip re-registration

Show Commands

Command Description
show sip trunk-registration shows registration status
show media-gateway summary shows media gateway channel status
show media-gateway channel 0/2.1 shows media gateway channel details
show qos map shows qos map
show flash lists contents of the flash
show run voice [verbose] lists running configuration for voice provisioning
show hosts shows the hosts table with timers
show event-history shows events and terminal commands typed
show interfaces shows interface details
show system shows uptime, firmware version and timing
show thresholds shows thresholds of T1 interfaces
show users shows current users
show voice quality-stats shows call information including lost packets
show voice quality-stats active shows call information including lost
show voice quality-stats active realtime shows call information including lost
show ip traffic TCP and UDP statistics