9133i doesn't register

You need to configure the Proxy Server and Proxy Port settings to point to your trixbox server.

As for the time server, I tried time.nist.gov, time.windows.com and myasterisk.dyndns.org. But it couldn't detect time either.

I put new firmware in /tftproot/ directory. Phone updated the firmware successfully. But stil didn't work. Now I've restored it to factory default settings. It doesn't say on display 'No Service' but doesn't register either and doesn't read config files any more from the /tftpboot/ directory.

Trixbox is not behind nat, but is on a public IP and is remote. I have 2 other IP phones and 3 softphones registered to it.

Now I am using the IP address instead of FQDN. But no difference.

The files are exactly the same between 1.4 and previous releases, that is the phone looks for aastra.cfg and .cfg; where is the mac address of the phone in uppercase.

The fact that the phone upgraded the firmware from the the trixbox tftp server, suggest that you've got TFTP connectivity and that the phone is downloading the aastra.cfg file. This is because the phone only checks for new firmware if it can successfully access the aastra.cfg file.

Have you created the .cfg files in /tftpboot? The aastra.cfg file only contains information about connecting to trixbox, the individual extension information for each phones is normally put in the .cfg file.

I don't know if you are using the end point manager, but I did notice that in trixbox 2.0 for some reason the end point manager didn't always write the .cfg file. I had to open the end points up again from the list and submit the information again to get it to write the files.

To prove one way or the other if the phone is connecting to the TFTP server you should run tail on the tftp logs why the phone is booting.

tail -f /var/log/atftpd.log

This will show you if the phone is downloading the files or not.

YES! The file name of the config file that you create and save needs to be in ALL CAPS.

Keep it like 00085D03F70F.cfg

This is the way it must be. Otherwise you will be fighting more then one problem and never get it resolved. Leave the file like this then we will know the problem is not the file name... and you can focus elswhere.

web gui settings will override the config file settings. The phones up load the config files, then override any setting by what was set on web gui. you need to factory reset the phone, reconfigure the phone to point to the tftp server and then try again

Thanks for all your help. But now I have following questions:

1. Why it tries to look for security.tuz file, where is this file located?

2. Purpose of TFTP is to control phones in a large deployment. If settings made through web UI or phone are over-riding TFTP's config settings, then what's the purpose of TFTP and how an administrator would change settings on, lets say, 50 phones.

3. When I enabled Force RFC2833 Out-of-Band DTMF setting, # started to work. In the dial plan it was not in quotes. I'll try that too. But what was happening in the beginning was that dialing *97# worked, but then entering password and then # didn't work. Was it because of RFC2833 Out-of-Band DTMF or because of it putting not being in quotes?

There are still few things I am trying to figure out on the phone, which I don't understand how are they working and/or not working. It was not easy to deal with this phone, but I am impressed by its excellent voice quality. This is the reason these days I am using only this phone. Purpose of asking these questions is to master its setup so in future I can confidently give them to my clients with voice quality guarantee.

Now my phone is working fine.

What I've learnt is following.

1. Web UI has problems in it.
2. Configuration file name should be capital.
3. Firmware file name should be changed to
.st, e.g. 9133i.st.
4. To use TFTP, don't touch anything in phone's UI or its web UI. This is what creates all the problems.
5. If phone's UI or web UI was used to make any settings, and later decided to use TFTP, first restore the phone to factory default settings and remove local configuration settings. This can be done in web UI. Make sure that aastra.cfg and .cfg files have all the required info. Then reboot the phone.
5. Use tail -f /var/log/atftpd.log command in linux to see if the phone is actually being served by the tftp server.
6. I also had to set DTMF methof to RTP and to enable Force RFC2833 Out-of-Band DTMF.

Why I say web UI has problems is because it acted funny ever since I got the phone, and I believe many things didn't work just because of this. For example, when I typed tftp server FQDN or IP, on rebooting the phone, everytime I saw last 2 letters or digits from the typed named were missing. For time server, it never updated its clock from the timer server, though it showed time server typed correctly.

Right now, what happened was that I put " marks around the typed dial plan. On reboot, there was no dial plan at all. Then I checked that in config file there were already quotes around the dial plan. I typed in the dial plan again without quotes, and on reboot again there was no dial plan. Obviously web UI settings were ignoring the configuration files' settings, and themselves were also not working. So I had to again restore it to factory default, so that config files can kick back it.

In short, phone works perfect once settings loaded from tftp. But web UI gives problems. I mostly use TFTP , so I am ok for now. But hopefully web UI will be fixed too by Aastra.

Otherwise, the ultimate feature, i.e. sound quality is wonderful and I've really liked this phone.

zishanov:

You only need the quote marks around the dial plan in the configuration file - you don't need to use them in the WebUI. They are needed to escape the '#' character in the configuration file, as this is the comment indicator for the configuration files only.

I've never heard of people having problems setting the TFTP server via the WebUI or TUI. Maybe there is something specific about the values you are trying to set (although I can't think what, unless you have a very long FQDN address), can you PM me the values you tried to use?

eoo:

Can you elaborate on the problems you are seeing using the WebUI? The only 1.4 web related issue that I know of is with long (85+ character) dial plans being truncated. Are there specific web pages that caused problems, or was it all of them? Was there a problem with a particular browser? Any info you can provide that can help isolate the problem, would be of great benefit.

Which timezone are you having problems with?

Note, you can only change the user password through the WebUI not the admin password. But I'm assuming you already knew that and it's the user you are having problems with?

I am also having issues in getting these phones to register. upgraded all phones to newest firmware without issue. Not sure about the global sip tab vs. the line 1 config, line 2 config, line 3 config etc...(9133i). My setup is no nat. Trixbox and all phones are on same lan. 3 9133i and 1 480i CT. Other phones Linksys spa922, spa921, Grandstream have no issue at all.

I went into the WEB UI and enter the user/password require by Trixbox for registration but ended up with a No service message. I generate the mac.cfg file with endpoint manager(Is there another way?).

I have 5 analog lines coming in. Just need it to be

L1 - 111-1111
L2 - 222-2222
L3 - 333-3333
L4 - 444-4444
L5 - 555-4444

Do I need to enter these actual phone numbers somewhere?

I had these extensions already in FreePbx

ext/password
1001/1001 - 9133i
1002/1002 - 9133i
1003/1003 - 9133i
1004/1004- 480i CT

Been through the forums and docs from Aastra.com. Maybe I haven't spend enough hours yet !

:-D

I did put my trixbox ip into the sip proxy and sip registrar already and set port 5060 for both. I am heading out for dinner now.!. Can't wait and try it out on the lines line1, line2. thanks

With Trixbox, all you need is to give an extension number and a password. In your case you'll add SIP extension. Then press submit and the red bar on the top. Once done, you'll click the extension from the extensions list, and in the settings screen, type 'no' in the 'qualify' field and 'yes' in the 'nat' field. You don't need to add/remove anything from sip.conf at this stage. The only thing you might need is to add a line 'insecure=invite' in sip.conf under the [general] parameters. But thats only to receive incoming calls on SIP trunks.

Once done, all you need is to setup aastra phone lines with the extension and passwords you just selected for your extensions.

To make it easier, at this stage keep the passwords same as extensions.

If you have trixbox on public IP, or forwarded to it can be accessed remotely, I can help you setup 2 extensions on it. I can also help you setup your one of your aastra using remote provisioning if you like. Then you can see what changes I've made and move on from there. If you want this, contact me at zeeshan@visionvoip.com with trixbox IP and MAC address of an aastra phone.

I need to add that I am not using any VOIP Provider. SO no sip calls coming in on trunks. Only from remote extensions. I Just got 5 incoming analog lines going into 5 fxo ports on a Sangoma a200d. Need to hard-code all 5 lines into their respective buttons on the Aastra's. The 5 lines are in a hunt group.

Already did that.. I'll just wait for an Aastra moderator to chime in. It just too bad. I was hoping to see a better written sample config files in the manual. I might just have to spend another 10-20 hours of reading. Anyway thanks again!

For such a popular phone, since nobody is having this issue, I guess I am just one of those dumb ones! :lol:

Have you used tftp? I think thats the only hope left for you.

Do setup-aastra on trixbox. It'll setup /tftpboot/ folder with a sample aastra config file in it. This file will be useless as it doesn't have anything in it.

Delete the three lines which are there in the aastra.cfg file and copy paste the following text in it. Save the file. In the phone's web UI, DO RESET it to factory default settings, reboot the phone and in the tftp section, give the IP address of your trixbox. And of course, in the following text, where I have 192.168.0.199, write your trixbox's IP address. Make sure you have extension 1111 with password 1111 in your trixbox. This is just for testing purposes to bring your phone online. Later you can change settings to whatever you like.

Please make sure you have qualify=no for the extension.

# DHCP Setting
# =============
dhcp: 1 # DHCP enabled.

# Time Server Settings
# =====================
time server disabled: 0
time server1: time.nist.gov

#sip outbound proxy: sbc.aastra.com
#sip outbound proxy port: 0 # a value of 0 enables SRV
# lookups for the address of
# the proxy.

sip rtp port: 10004
#-----------------------------------------------------------------

# Configuration Server Settings
# = = = = = = = = = = = = = =
download protocol: TFTP
tftp server: 192.168.0.199

#----------------------------------------------------------------
# Dial Plan Settings
# ====================
sip digit timeout: 3
sip dial plan: "x+#|xx+*"
#-----------------------------------------------------------------
# General SIP Settings
# ======================
sip session timer: 30
sip transport protocol: 0 # use UDP (1), TCP (2) or both (0)

# Global SIP User Settings
# ==========================
#
# sip screen name: Your Name # the name display on the phone's screen
# sip user name: 1111 # the phone number
# sip display name: Your Name
# sip vmail: *97
# sip auth name: 1111
# sip password: 1111
# sip mode: 0 # line type:
# 0 - generic,
# 1 - BroadSoft SCA line
# 2 - Nortel line
# sip proxy ip: 192.168.0.199
# sip proxy port: 5060
# sip registrar ip: 192.168.0.199 # IP address or FQDN of registrar
# sip registrar port: 5060 # as proxy port, but for the registrar
# sip registration period: 60 # registration period in seconds

# Per-line SIP Settings
# =======================
# configure line 1 as your first extension

sip line1 screen name: Zeeshan-VisionVoIP
sip line1 user name: 1111
sip line1 display name: Zeeshan-VisionVoIP
sip line1 auth name: 1111
sip line1 password: 1111
sip line1 mode: 0
sip line1 proxy ip: 192.168.0.199
sip line1 proxy port: 5060
sip line1 registrar ip: 192.168.0.199
sip line1 registrar port: 5060
sip line1 registration period: 300

# sip line3 mode: 1
# sip line3 vmail: *78
# configure line 5 (a soft key line) as an ordinary line
# of a test server
# sip line5 screen name: Test 1
# sip line5 user name: 5551001
# sip line5 display name: Test 1
# sip line5 auth name: 5551001
# sip line5 password: 5551001
# sip line5 mode: 0
# sip line5 proxy ip: 10.50.10.102
# sip line5 proxy port: 5060
# sip line5 registrar ip: 10.50.10.102
# sip line5 registrar port: 5060
# sip line5 registration period: 60
#-----------------------------------------------------------------
# Programmable Key Settings
# ===========================
# Programmable keys can be set either server wide or unique to each
# phone.
# Setting programmable keys as line/call appearances should be done
# in the ".cfg" file, since these are unique to each phone.
# Notes:
#
# There are a maximum of 7 programmable keys that can be configured
# on the 9133i phone, and only 2 on the 9112i phone. These can be
# set up through either of the 2 configuration files, depending on
# whether this is to be server wide ("aastra.cfg") or phone
# specific (".cfg"). Each prgkey needs to be numbered from
# 1 - 7 (or 1 -2 on the 9112i), for example "prgkey2 type:
# speeddial". Programmable keys can be set up as speeddials or as
# additional call/line appearances or as feature keys and have a
# type, value and line associated with it as seen here in the
# default programmable settings.
# PRGKEY TYPES: "line", "speeddial", "blf", "list", "dnd"
# PRGKEY VALUE: If prgkey type is a speeddial, any DTMFs (from
# 0 - 9, *, "#") or a comma (,) for 500ms pause and
# 'E' for On-hook can be set for the value.
# If prgkey type is blf it is the extension you want
# to monitor.

did the setup-aastra got the file aastra.cfg .. how do you open it?
I used notepad and it came out weird. You mentioned not to use the web-gui, or phone gui, after I copy and pasted your config. then I should set phone to factory default, reboot all my phones and presto right!?

Zishanov

Thanks for saving me a few more hours of reading! I got one of my extensions to register. By successfully registering one phone, it answered alot of my previous questions but one more which I am going to have to find out at the client site on Saturday. I don't have 5 fxo lines coming in to test but I do have the general idea of how it works. It will be down hill from here for sure. First off, I didn't use tftp(only 4 Aastras to config) . I went into the Web-Gui and put in the credentials. here is what I found out so far with testing for newbies that are in the same boat and to save them hours of reading.

I am really confused by all the credentials under Basic Sip Configuration Settings.(other phones just need login name/password) I started with the obvious until I get it to make a call.

1. I put in credentials for authtentication and password(not enough)
2. Added a "Screen Name"-1001- and rebooted(much better after phone booted up. Saw 1001 on the display. BUT couldn't make any calls or *43 or anything. Basically STILL not register. O.K.
3. Took out screen name and leave blank and added 1001 to "Phone Number" and rebooted. That did the trick!!. So Phone Number is a critical field. !!!!!! After rebooting the display showed 1001 on the display. I noticed that there are 2 lines available. IF I kept the 1001 on the Screen the after I rebooted the phone I will have 2 instances of my extensions displayed. One on top of each other. But the Screen Name credentials are not necessary to register and make phone calls.

I don't have access to analog lines at this point so I still don't know under Phone Number if I should input my real analog phone lines. But experimentation will tell tomorrow.

I also notice that seem like the line1, line2 , line 3 in the config web page doesn't really refer to the physical buttons on the phones. I went into line2 and of course it was blank. But on my physical line 2 and line 3 I can still hear dial-tone and make phone calls. Hm...hmm..hmm

I also notice that if I input the config file for the credentials on the sip global..It will input go ahead and input all the data for line1, line2, line3, line 4, line 5 , line 6 .(I mean the web-page gui not physical line buttons)
It will not input the credentials for the screen name or the phone number. Weird. I swear that I saw that during my testing. But a second time, I couldn't duplicate my results.

The result was that entering all necessary credentials on the sip global page, I get it to work also. Line1, line2, line3 all was blank. Darn.. I thought that I was on to something.! :lol:

_____________________________________________
my plan of deployment is as follows:

Make a ring group and call it "receptionist"and put all those extensions into it
1001
1002
1003
1004

In the inboud route point it to this ring group. That should ring all phones at the same time. The should take care of all those incoming lines. Welcome comments on best way to accomplish my scenario.

Aastra

Thanks for responding. But my questions remains. I am more interested in getting INCOMING questions answered because it is suppose to be a small call center. Excuse me again, I don't know if anybody is understanding my rambling on and on so I will need to re-ask my concerns in my previous post.

1. So you're saying that the Aastra Phones doesn't have per fxo line(number) map to physical button??
2. So what does the Aastra line button(the green ones) do?? L1, L2, L3
Those are obviously hardcoded from factory for Line(fxo) designations.
I don't believe that I can even program speed-dail into those buttons if I wanted to. As I said I have 5 lines.

3. If People start calling the main phone line (111-1111) and talking while someone else call in and got roll over to line 2(222-2222). 3rd person call in got rolled over to line3(333-3333). 4th person call in..which button will it roll to.?? which bring me to my question what are those line1, line2, line3, line4, mapped to??

4. which bring me to my question? Is it doe-able to map the buttons on the right hand side to fxo so that calls can flow there? IF yes, then
which button correspond to what. For simplicity I want the last 2 buttons on the right panel above the Mute button on the 9133i . That will be for L4 and L5. What button does those 2 buttons correspond to?

That's all for now, appreciate any input.

Zishanov: Many of your comments are true with regards to the user needing to learn more about Aterisk, and how it works and why. HOWEVER, as a VERY experienced user and installer, I can tell you that the 9133i is NOT difficult to configure. It is often impossible, and quite simply pot luck. EXAMPLE: installed 16 Phones sucessfully. 36 would not install. All returned. At another install none of the phones would install, then 3 would randomly register. I know EVERY trick with these phones. They are just NO good. It should not be this difficult to setup an endpoint. Period. My advice to anyone reading this is simple. DO NOT BUY THESE CRAPPY PHONES. YOU WILL BE IN TEARS. Whilst the voice quality is good and the phone functions well, you can get random errors and problems that you will NEVER find the solution to. Put simply, the firmware is so bad and buggy, that it simply is not worth using.

GENERAL RULE: Don't buy aasta phones. If you have sell them. If they work for you, feel blessed. And if your an installer, never install them in a commercial environment. I have probably installed more of these handsets than anyone else I have ever heard of. I'm replacing them at the rate of 3 a week at my cost.

Buy a SNOM, Grandstream, CISCO, Linksys - whatever just NOT an aastra.

Until aastra understands that the programming is BAD, no body should buy these phones, period. Yes I know, they look good, reasonable price, and some people including myself have used them without problem - the exception - not the rule.

I have installed almost every sip phone known to man. These with out doubt, will hurt you, and make you cry. Lots!

So take heed: If you have one and it doesnt work try to return it. Don't waste your time. The solutions above sometimes work, sometimes don't. Its just luck. If you cant return the phone, bin it, or ebay it. Don't waste hours trying to fix a problem that in all likelyhood, you will never find a solution to.

Some aastra phones are better than others. By and large they are very good phones with excellent voice quality and good features. Shame that they have such badly written firmware.

I'm sure some will slate me - But I have a binful of them - not selling them as I don't think it's fair. If I cant get them to work - Joe Bloggs certainly can't.

AASTRA - if your reading this Take note. Get your act together, and fix your damn phones. You have a great many installers including myself, that loathe you. You dont listen. Your support sucks. You have great phones with BAD firmware.