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UTStarcom iAN-02EX firmware

Started by mloewen, December 18, 2018, 12:11:27 AM

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mloewen

I recently acquired a UTStarcom iAN-02EX ATA, to connect my WE 500D rotary phone to my Asterisk PBX.  The details are:

Board ID:    U53V004.00.00_UTSTARCOM
Firmware Version:    V1.8.2.15a-Asterisk

After much fighting with this unit, I've come to the conclusion that this particular firmware version only works with MGCP, and not with SIP.  Configured for SIP, it will ring when called, but it never dials out: it tries to use MGCP to dial, no matter what SIP settings are used.  Configured for MGCP (and a custom extension in Asterisk), it works for both dialing and answering.

From other posts in this forum, I've seen mention of much later firmware versions for this ATA, but they've all disappeared from the net by now. Does anyone have a copy of V1.8.2.51b, or something of that vintage?  PM me, if you like.

Yes, I'm aware of other ATAs that support pulse dialing, such as the Grandstream HT502.  I have an HT502, but it seems to be finicky in dialing.  Sometimes it gets the dialed numbers wrong, whereas the iAN-02EX gets it right every time.

Thanks for any help.

markosjal

#1
The UTStarcom_AzaCall came from the Aza Call branded units


I run V4.8.2.50b which I have attached last

If you have a Lingo branded box I think you will need the Lingo unlocked firmware, but you could try the 50b version here.The most important thing on the Lingo version is you may not be able to use both phone ports and as long as you remove the provision settings and MAKE A NOTE OF THE PASSWORD, all should be fine.

For a Lingo locked adapter you can try this for firmware version V2.8.2.63b
Disconnect WAN cable.

with computer connected to LAN port

Do a factory reset while keeping it off the internet.

Then:

user
ph3taswe

supervisor (try both)
$KunK$pit
GoHaWk3Y3$

After that, login and remove provisioning and change the password.



To reset

Disconnect WAN port

Power up the ATA. Plug a handset into Line1 and dial *#26845#

default user and pass supervisor supervisor (or supervisor password?)

some may be supervisor / wre4wecr (azacall?)
Phat Phantom's phreaking phone phettish

markosjal

#2
If you need an alternative to SIP protocol with asterisk ,MGCP configuration for Asterisk 13 and UTStarcom IAN02EX with both on same LAN. Tested with UTStarcom Firmware 4.8.2.50b and asterisk 13

THIS ONLY APPLIES TO MGCP PROTOCOL, NOT SIP. SIP IS THE PREFERRED PROTOCOL FOR THESE ATAS ON ASTERSK. I PROVIDE THE MGCP INFORMATION ONLY AS AN ALTERNATE METHOD. SCROLL DOWN TO FIND SIP PROTOCOL CONFIGURATION.

Gateway

Give the gateway a fixed IP address:
See attachment 1

Set protocol to MGCP :
See attchement 2

Set MGCP as shown :
See attachment 3 and setting gatway doman and call agent address to the IP of your asterisk



Asterisk


Of course if you use a GUI based distro of asterisk like Issabel or FreePBX you need to put some of these these in different files

mgcp.conf (just add to end of file , making changes where necessary) :

[192.168.1.49] ;UTStarcom fixed IP
host = 192.168.1.49  ;UTStarcom fixed IP
context = from-internal  ;your context fr internal callers same as those extensions in sip.comf
directmedia = no
canreinvite=no
wcardep = *
callgroup=1
callerid="MGCP1" <5921> ; This will apply to all lines "line" after until callerid is defined again set to caller ID name and number
line => aaln/1
mailbox=5921 ; mailbox on extension
threewaycalling=yes
callwaiting=yes
transfer=yes
cancallforward = yes
callreturn = yes
dtmfmode=rfc2833
callgroup=1
context = from-internal
callerid="MGCP2" <5922>   ;set to caller ID name and number
line => aaln/2
mailbox=5922 ; mailbox on extension
threewaycalling=yes
callwaiting=yes
transfer=yes
cancallforward = yes
callreturn = yes
dtmfmode=rfc2833


This last bit associates an extension number with a particular MGCP device and channel and should be added in the from-internal or whatever other local context yiu use.

extensions.conf SEE NEXT POST FOR UPDATE TO GET PROPER RING BACK AND BUSY SIGNAL
Now because I used 2 extensions only 5921 and 5922 I used astersk to eliminate the first three digits with "EXTEN:3". You must also set the IP address of the UTStarcom as this is how we identify it

exten => _592[12],1,Dial(MGCP/aaln/${EXTEN:3}@192.168.1.49)
exten => _592[12],2,Hangup()


Alternate code for extensions.conf SEE NEXT POST FOR UPDATE TO GET PROPER RING BACK AND BUSY SIGNAL

exten => _5921,1,Dial(MGCP/aaln/1@192.168.1.49)
exten => _5921,2,Hangup()

exten => _5922,1,Dial(MGCP/aaln/2@192.168.1.49)
exten => _5922,2,Hangup()



Problems noted:

Not hearing ringback when calling SIP extensions. (see work-around posted below)

Unable to determine how to enable disable VMWI . 1 checks for Voice Mail 2 does not.

Phat Phantom's phreaking phone phettish

markosjal

#3
for anyone interested or who may later find this, I have found a work-around for the ringback problem and extension to extension calling.

THIS ONLY APPLIES TO MGCP PROTOCOL, NOT SIP. SIP IS THE PREFERRED PROTOCOL FOR THESE ATAS ON ASTERSK. I PROVIDE THE MGCP INFORMATION ONLY AS AN ALTERNATE METHOD. SCROLL DOWN TO FIND SIP PROTOCOL CONFIGURATION.

Of course if you use a GUI based distro of asterisk like Issabel or FreePBX you need to put these in different files

I created a custom music on hold that plays a ringback the "m(ringback)" means play Music On hold (foldername) . I also created "m(engaged)" for a busy signal.

copy ringback files from attached zip to /var/lib/asterisk/ringback changing owner to asterisk:asterisk
copy busy files from attached zip to /var/lib/asterisk/engaged changing owner to asterisk:asterisk

If you want old style tones or other than N american you can change the files later and do "moh reload" from asterisk conole, but a reboot may be necessary. It is best to use the provided files for testing.


what is below, goes in to musiconhold.conf , just below [default]

[ringback]
mode=files
directory=/var/lib/asterisk/ringback ; full path to ringback files

[engaged]
mode=files
directory=/var/lib/asterisk/engaged ; full path to ringback files


I changed extensions.conf to the following to accommodate where ringback worked and where it did not work

;fake ringback and busy

; dummy extension rings for 30 seconds while we play busy (engaged) tone
exten => 8888,1,Wait(30)
exten => 8888,2,Hangup()

;MGCP to MGCP dialing
exten => _592X/_592X,1,Dial(MGCP/aaln/${EXTEN:3}@192.168.1.49,,m(ringback))
exten => _592X/_592X,2,Dial(Local/8888@from-internal,,m(engaged))
exten => _592X/_592X,3,Hangup()

;MGCP to SIP dialing
exten => _59[01]X/_592X,1,Dial(SIP/${EXTEN},,m(ringback))
exten => _59[01]X/_592X,2,Dial(Local/8888@from-internal,,m(engaged))
exten => _59[01]X/_592X,3,Hangup()

;Real ringback
;SIP to MGCP
exten => _592X/_59[01]X,1,Dial(MGCP/aaln/${EXTEN:3}@192.168.1.49)
exten => _592X/_59[01]X,2,Hangup()

;SIP to SIP
exten => _59[01]X/_59[01]X,1,Dial(SIP/${EXTEN},,Tr)
exten => _59[01]X/_59[01]X,2,Hangup()


Alternate ringback code (might be easier to understand, but longer)


;fake ringback and busy
; dummy extension rings for 30 seconds while we play busy (engaged) tone
exten => 8888,1,Wait(30)
exten => 8888,2,Hangup()
;MGCP to MGCP dialing
exten => _5922/_5921,1,Dial(MGCP/aaln/2@192.168.1.49,,m(ringback))
exten => _5922/_5921,2,Dial(Local/8888@from-internal,,m(engaged))
exten => _5922/_5921,3,Hangup()

exten => _5921/_5922,1,Dial(MGCP/aaln/1@192.168.1.49,,m(ringback))
exten => _5921/_5922,2,Dial(Local/8888@from-internal,,m(engaged))
exten => _5921/_5922,3,Hangup()

;MGCP to SIP dialing
exten => _59[01]X/_592X,1,Dial(SIP/${EXTEN},,m(ringback))
exten => _59[01]X/_592X,2,Dial(Local/8888@from-internal,,m(engaged))
exten => _59[01]X/_592X,3,Hangup()

; end fake ringback

;Normal ringback
exten => 5922/_59[01]X,1,Dial(MGCP/aaln/2@192.168.1.49)
exten => 5922/_59[01]X,2,Hangup()

exten => 5921/_59[01]X,1,Dial(MGCP/aaln/1@192.168.1.49)
exten => 5921/_59[01]X,2,Hangup()

exten => _59[01]X/_59[01]X,1,Dial(SIP/${EXTEN},,Tr)
exten => _59[01]X/_59[01]X,2,Hangup()


you can add your voice mail if needed

Reboot the asterisk and tes
Phat Phantom's phreaking phone phettish

markosjal

This is the SIP configuration of UTStarcom IAN-02-EX for asterisk. This is the recommended configuration. If you have issues with this then you can try the alternate MGCP configuration posted above.

This assumes you have set up your extensions on asterisk. There are several web GUIs in use and Vanilla asterisk, so you need to figure that part out for yourself.  in this example we use 5901 and 5902

Connect your computer to the LAN Port of the IAN-02EX

in a web browser open 172.25.25.1
You should be promted with username and password. (see other post)

From the menu pane on the left we will work under the VoIP menu and set items as seen in images.

of course you will need to change username, login ID, and displayname to conform to the extensions on your asterisk. Set password to the password of your extension defined in asterisk. You will need to edit IP address fields in VoIP  > SIP to the IP address of your asterisk.

Phat Phantom's phreaking phone phettish

dsk

#5
Now I need help again :'(
I should change the phone-numbers on my UTStarcom IAN-02-EX and lost the connection.
I have done a full reset but still problems to reach it on the network.
I may log inn on the LAN side (but gets no connection with the WAN).


markosjal

Sorry for my delayed reply....

I think you need to login on the LAN side then once there, enable the login over WAN
Phat Phantom's phreaking phone phettish

dsk

Did not work out, and it even struggeled to be connected to the net, so it's gone.

The Grandstream does the job for me now. 

Moving to a smaller apartment so one ATA for the exchange, and one for my payphones that need reverced polarity on answer is all I need.

Struggeling to find a good home for may electromechanical PAX, It is to noisy and large for an apartment. :-)