Starting out: Aastra 9133i and Asterisk

My, this is a steep learning curve!

I have a system running Asterisk 1.2.10 on Centos 4.3 and have tried to connect my Aastra 9133i phones running firmware 1.4.0.1048. All I get so far is “No Service” on the LCD. I hope that someone can point me in the right direction.

My aastra.cfg file looks like this:

[code]; GLOBAL CONFIGURATION FOR ALL AASTRA PHONES

; #### Network ###

dhcp: 1

; ### Password ###

admin password: 22222
user password: 22222
options password enabled: 1

; ### Web UI ###

web interface enabled: 1

; ### Configuration Server ###

download protocol: tftp
tftp server: 192.168.4.2
auto resync mode: 1
auto resync time: 00:00

; ### ToS/DSCP ###

; ### VLAN ###

; ### NAT ###

; ### Time Server ###

time server disabled: 0
time server1: 192.168.0.3
date format: 1
time format: 1

; ### SIP local dial plan ###

sip dial plan: “0XXX+|1X”

; ### SIP Registration ###

; ### SIP Basic, Global ###

sip mode: 0
sip registrar ip: 192.168.4.7

; ### SIP Basic, Per Line ###

; ### Advanced SIP ###

; ### RTP, Codec, DTMF globals ###

; ### DTMF Per Line ####

; ### Silence Suppression ###

; ### Voicemail ###

; ### Directory ###

; ### Caller List ###

; ### Caller Forward ###

; ### Missed Calls Indicator ###

; ### XML ###

; ### Ring Tone and Tone Set Globals ###

ring tone: 0
tone set: UK

; ### Ring Tone per line ###

; ### Stuttered Dial Tone ###

; ### Call Waiting Tone ###

; ### Priority Alert ###

; ### Language ###

language: 0 ; English

; ### Suppress DTMF Playback ###

; ### Intercom and Auto-Answer ###

; ### Audio Gain Adjustment ###

; ### Direct Call Pickup ###

direct call pickup: 1

; ### BLF Subscription ###

; ### Hard Keys ###

; ### Soft Keys ###

; ### Programable Keys ###

; ### Advanced Params ###
[/code]

My .cfg files look like this:

[code]; CONFIGURATION FOR PHONE 00085D****** - Denise

; ### Network ###

; ### Password ###

; ### Web UI ###

; ### Configuration Server ###

; ### ToS/DSCP ###

; ### VLAN ###

; ### NAT ###

; ### Time Server ###

; ### SIP Local Dial Plan ###

; ### SIP Registration ###

; ### SIP Basic, Global ###

sip screen name: Denise C
sip user name: aastra10
sip display name: Denise C
sip auth name: aastra10
sip password: ********

; ### SIP Basic, Per Line ###

; ### Advanced SIP ###

; ### RTP, Codec, DTMF globals ###

; ### DTMF Per Line ####

; ### Silence Suppression ###

; ### Voicemail ###

; ### Directory ###

; ### Caller List ###

; ### Caller Forward ###

; ### Missed Calls Indicator ###

; ### XML ###

; ### Ring Tone and Tone Set Globals ###

; ### Ring Tone Per Line ###

; ### Stuttered Dial Tone ###

; ### Call Waiting Tone ###

; ### Priority Alert ###

; ### Language ###

; ### Suppress DTMF Playback ###

; ### Intercom and Auto-Answer ###

; ### Audio Gain Adjustment ###

; ### Direct Call Pickup ###

; ### BLF Subscription ###

; ### Hard Keys ###

; ### Soft Keys ###

; ### Programable Keys ###

; ### Advanced Params ###
[/code]

And my sip.cfg file is the default with these lines tagged on the end:

[aastra10] type=friend host=dynamic defaultip=192.168.4.6 secret=******** dtmfmode=rfc2833 mailbox=1000 context=sip callerid="Denise"

I don’t see a ‘username=aastra10’ in your sip.conf. That’s a good start. I have the 9133i working VERY well with latest Asterisk!

i think you need to also define the ‘sip proxy ip’…

I assume that this is the IP of the Asterisk box - correct?

I assume that this is the IP of the Asterisk box - correct?[/quote]

yup

Sorry guys, I’m still not getting anywhere.

Here is my current aastra.cfg

[code];
; Global Configuration for all Aastra phones
;

; #### Network ###

dhcp: 1

; ### Password ###

admin password: 22222
user password: 22222
options password enabled: 1

; ### Web UI ###

web interface enabled: 1

; ### Configuration Server ###

download protocol: tftp
tftp server: 192.168.4.2
auto resync mode: 1
auto resync time: 00:00

; ### ToS/DSCP ###

; ### VLAN ###

; ### NAT ###

; ### Time Server ###

time server disabled: 0
time server1: 192.168.0.3
date format: 1
time format: 1

; ### SIP local dial plan ###

sip dial plan: “0XXX+|1X”

; ### SIP Registration ###

; ### SIP Basic, Global ###

sip mode: 0
sip registrar ip: 192.168.4.7
sip proxy ip: 192.168.4.7

; ### SIP Basic, Per Line ###

; ### Advanced SIP ###

; ### RTP, Codec, DTMF globals ###

; ### DTMF Per Line ####

; ### Silence Suppression ###

; ### Voicemail ###

; ### Directory ###

; ### Caller List ###

; ### Caller Forward ###

; ### Missed Calls Indicator ###

; ### XML ###

; ### Ring Tone and Tone Set Globals ###

ring tone: 0
tone set: UK

; ### Ring Tone per line ###

; ### Stuttered Dial Tone ###

; ### Call Waiting Tone ###

; ### Priority Alert ###

; ### Language ###

language: 0 ; English

; ### Suppress DTMF Playback ###

; ### Intercom and Auto-Answer ###

; ### Audio Gain Adjustment ###

; ### Direct Call Pickup ###

direct call pickup: 1

; ### BLF Subscription ###

; ### Hard Keys ###

; ### Soft Keys ###

; ### Programable Keys ###

; ### Advanced Params ###
[/code]

one of the .cfg

[code];
; Configuration for Aastra phone 0008******** (Denise)
;

; ### Network ###

; ### Password ###

; ### Web UI ###

; ### Configuration Server ###

; ### ToS/DSCP ###

; ### VLAN ###

; ### NAT ###

; ### Time Server ###

; ### SIP Local Dial Plan ###

; ### SIP Registration ###

; ### SIP Basic, Global ###

sip screen name: Denise C
sip user name: aastra10
sip display name: Denise C
sip auth name: aastra10
sip password: ********

; ### SIP Basic, Per Line ###

; ### Advanced SIP ###

; ### RTP, Codec, DTMF globals ###

; ### DTMF Per Line ####

; ### Silence Suppression ###

; ### Voicemail ###

; ### Directory ###

; ### Caller List ###

; ### Caller Forward ###

; ### Missed Calls Indicator ###

; ### XML ###

; ### Ring Tone and Tone Set Globals ###

; ### Ring Tone Per Line ###

; ### Stuttered Dial Tone ###

; ### Call Waiting Tone ###

; ### Priority Alert ###

; ### Language ###

; ### Suppress DTMF Playback ###

; ### Intercom and Auto-Answer ###

; ### Audio Gain Adjustment ###

; ### Direct Call Pickup ###

; ### BLF Subscription ###

; ### Hard Keys ###

; ### Soft Keys ###

; ### Programable Keys ###

; ### Advanced Params ###
[/code]

My sip.conf

[code];
; SIP Configuration for Asterisk
;

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[authentication]

[10]
username=aastra10
type=friend
host=dynamic
defaultip=192.168.4.6
secret=********
dtmfmode=rfc2833
mailbox=1000
context=sip
callerid=“Denise”

[11]
username=aastra11
type=friend
host=dynamic
defaultip=192.168.4.8
secret=********
dtmfmode=rfc2833
mailbox=1000
context=sip
callerid=“Janet”

[12]
username=aastra12
type=friend
host=dynamic
defaultip=192.168.4.9
secret=********
dtmfmode=rfc2833
mailbox=1000
context=sip
callerid=“Barbara”
[/code]

and my extensions.conf

[code];
; Extensions Configuration for Asterisk
;

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp

[sip]
exten => 10,1,Dial(SIP/10,20)

exten => 11,1,Dial(SIP/11,20)

exten => 12,1,Dial(SIP/12,20)
[/code]

do a sip debug on *, then reboot the phone. Paste here whatever you get.

Also what FW ver are you running? If anything but the latest (1.4 i think) then upgrade.

After you upgrade reset phone to factory defaults with the keypad, then try this config again. (that helped for some reason with older firmwares…)

The firmware is up to date (version 1.4.0.1048)

I tried a factory reset (via the web interface) but that made no difference.

Here are the results of the debug session:

[code]localhostCLI> sip debug ip 192.168.4.6
SIP Debugging Enabled for IP: 192.168.4.6
localhost
CLI>
<-- SIP read from 192.168.4.6:5060:
REGISTER sip:192.168.4.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.6;branch=z9hG4bK81845f7ba
Max-Forwards: 70
Content-Length: 0
To: Denise sip:Denise@192.168.4.7
From: Denise sip:Denise@192.168.4.7;tag=398d1e92521b11a
Call-ID: 2ce7332e2f948178bea5d7c7d35ecc50@192.168.4.6
CSeq: 1934522619 REGISTER
Contact: Denise sip:Denise@192.168.4.6
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
User-Agent: Aastra 9133i/1.4.0.1048 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26

— (12 headers 0 lines)—
Using latest REGISTER request as basis request
Sending to 192.168.4.6 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.4.6:5060:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.4.6;branch=z9hG4bK81845f7ba;received=192.168.4.6
From: Denise sip:Denise@192.168.4.7;tag=398d1e92521b11a
To: Denise sip:Denise@192.168.4.7;tag=as7d1eef6f
Call-ID: 2ce7332e2f948178bea5d7c7d35ecc50@192.168.4.6
CSeq: 1934522619 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:Denise@192.168.4.7
Content-Length: 0


Aug 28 12:39:19 NOTICE[2348]: chan_sip.c:11045 handle_request_register: Registration from ‘Denise sip:Denise@192.168.4.7’ failed for ‘192.168.4.6’ - Username/auth name mismatch
Scheduling destruction of call ‘2ce7332e2f948178bea5d7c7d35ecc50@192.168.4.6’ in 15000 ms
Destroying call '2ce7332e2f948178bea5d7c7d35ecc50@192.168.4.6’
localhost*CLI>[/code]I tried making every sort of name the same but that still made no difference. I have to get this running by next weekend - help!!!

sip user name has to be the same thing as in the [definition] thing of sip.conf. also try setting all the sip names (username/displayname/etc) to be Denise. I think it’s case sensitive

Wooo Hooo - I’ve made my first internal call!

Thanks IronHelix, that was exactly the tip I needed. In fact, I’ve removed the “username=Denise” line completely and it still works fine.

Now I’ll experiment to try and strip things down to the bare minimum and then I’ll work on the outside line.

Congrats! FYI you should be able to change the display name (maybe its screen name, one of those) without a bad effect, as one is just printed on the display and nothing else.

The 9133 is a great phone. Good luck!

We’re getting there - slowly!

I can make incomming calls and, via a queue, I can even ring several phones at once. However, the caller hears nothing while these phones are ringing. I’m not sure what is going on.

-- Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx: > requested format = ulaw, > requested prefs = (), > actual format = gsm, > host prefs = (), > priority = mine -- Executing Answer("IAX2/xxx.xxx.xxx.xxx:4569-2", "") in new stack -- Executing Queue("IAX2/xxx.xxx.xxx.xxx:4569-2", "Qoffice") in new stack -- Started music on hold, class 'default', on channel 'IAX2/xxx.xxx.xxx.xxx:4569-2' -- Called SIP/10 -- Called SIP/11 -- Called SIP/12 -- SIP/11-09bfb280 is ringing -- SIP/10-09bf5d70 is ringing -- SIP/12-09c00790 is ringing Aug 28 22:16:50 NOTICE[2324]: res_musiconhold.c:511 monmp3thread: Request to schedule in the past?!?! -- Stopped music on hold on IAX2/xxx.xxx.xxx.xxx:4569-2 == Spawn extension (gradwell-in, xxxxxxxxxxx, 2) exited non-zero on 'IAX2/xxx.xxx.xxx.xxx:4569-2' -- Hungup 'IAX2/xxx.xxx.xxx.xxx:4569-2' Aug 28 22:17:10 NOTICE[2324]: res_musiconhold.c:511 monmp3thread: Request to schedule in the past?!?!

I’ve even tried creating a MusiconHold extension but when I call it I hear nothing.

Ideally, the user would just like the caller to hear a simple ring tone. Is there one available?

first, your MOH is broken, need to fix that. in asterisk source dir do ‘make mpg123’, which will get you set up.

if you add the ‘r’ flag to Queue(), it will play ringing instead of MOH…

Thanks for all this help IronHelix. You’re much more use than the book Building Telephone Systems with Asterisk! It’s riddled with mistakes and IMHO neither a reference nor a ‘how to’ book. It falls some way between the two stools.

I’ve got the ring tone working now - thanks.

And I’ve made mpg123 as you suggested but MOH is still quiet. I’ve created an extension, 19, that simply connects to it using this:

But when I call it I hear nothing and see this:

-- Executing MusicOnHold("SIP/10-087b1ea0", "default") in new stack -- Started music on hold, class 'default', on SIP/10-087b1ea0 Aug 29 11:53:27 WARNING[3042]: file.c:512 ast_openstream_full: File /var/lib/asterisk/mohmp3/fpm-sunshine does not exist in any format Aug 29 11:53:27 WARNING[3042]: res_musiconhold.c:227 ast_moh_files_next: Unable to open file '/var/lib/asterisk/mohmp3/fpm-sunshine': No such file or directory -- Stopped music on hold on SIP/10-087b1ea0 == Spawn extension (office, 19, 1) exited non-zero on 'SIP/10-087b1ea0'

The files are there, in mp3 format, but Asterisk can’t see them.

My musiconhold.conf file is like this:

[code];
; musiconhold.conf
; Music on Hold configuration for Asterisk
;

[default]
mode=mp3
directory=/var/lib/asterisk/mohmp3
random=yes[/code]

Are you using the ztdummy module or do you have some digium/spinoff cards in the asterisk box?

There are no Digium (or similar) cards in the box. I’m not sure about the ztdummy module - how do I look for that?

do you have zaptel installed at all?

Also try the book Asterisk: The Future of Telephony. I’ve heard good things about it.

As far as I am aware, I’m not running Zaptel.

And here’s my modules.conf (unmodified since installation)

[code];
; Asterisk configuration file
;
; Module Loader configuration file
;

[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using ‘preload’. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those configuration
; files are initialized.
;
; An example of loading ODBC support would be:
;preload => res_odbc.so
;preload => res_config_odbc.so
;
; If you want, load the GTK console right away.
; Don’t load the KDE console since
; it’s not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don’t load it.
;
noload => app_intercom.so
;
; The ‘modem’ channel driver and its subdrivers are
; obsolete, don’t load them.
;
noload => chan_modem.so
noload => chan_modem_aopen.so
noload => chan_modem_bestdata.so
noload => chan_modem_i4l.so
;
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
;noload => chan_oss.so
;
; Module names listed in “global” section will have symbols globally
; exported to modules loaded after them.
;
[global][/code]

as far as you are aware? is this a trixbox install?

if you compiled asterisk from source adn didnt install zaptel too, then you dont have it. zaptel is a separate package which must be installed on its own…

In that case, no, I don’t have Zaptel installed. I compiled Asterisk from the source. Does Zaptel make a difference to MOH?