I have two questions to see if I will use MTE for my Hosted Solution. So far I like it, but I cannot get any phone to register (no service) and I am baffled.
I am using 9133i and 480i.
1. All I am doing is configuring the extension (with MAC) in the PBX Manager and then going into the phone and placing the TFTP settings. It appears to grab the cfg file, but the phone will not work. I have tried three phones. Is there something I'm missing?? TFTP is runnign and cfg files are there and loaded.
2. What phones do people like (other than the ones mentioned above) with this PBX that is clean, sleek and if possible, gives BLF?
Thanks!!!
Xlite Appears to work fine.
Xlite Appears to work fine. I wonder why my Aastras will not work. Manuals say it's self provisioning with MAC but no such luck hear. Really looking for a phone I can Private Label and Aastra lets you do that. Any others?
given enough capitol i know
given enough capitol i know polycom will. 3com is rebranding polycom and so is Adtran. I would say your next experiment is to check the TFTP logs to ensure that the correct files are being downloaded. If the are, then manually inspect your config files to make sure they look correct.
I have about 12 55i Aastras
I have about 12 55i Aastras that I configured initially with PBX Mgr. If I remember correctly it worked. But, I eventually tuned the config files manually. I prefer Aastra's provisioning mechanism over Polycom's, but have decided to use Polycom going forward for our customers because of the overall quality of the phone.
Are you sure the phones are picking up the config info? You should see your settings in the phone's web GUI.
Here is my aastra.cfg file. Notice the custom codec settings. This line sets the phone up to do g.711 and g.729.
# aastra registration
backlight mode: 2
sip proxy ip: x.x.x.x
sip proxy port: 5060
sip registrar ip: x.x.x.x
sip registrar port: 5060
sip registration period: 30
sip registration retry period: 30
sip registration timeout retry: 30
sip registration renewal timer: 20
sip use basic codecs: 1
auto resync mode: 3
auto resync time: 00:00
sip customized codec: payload=0;ptime=10;silsupp=off,payload=18;ptime=20;silsup=off
time server1: rolex.usg.edu
time server2: timex.usg.edu
time server disabled: 0
time zone name: US-Eastern
time zone code: EST
dst config: 3
download protocol: FTP
ftp server: x.x.x.x
ftp username: xxxxxxxx
ftp password: xxxxxxxx
sip dial plan terminator: 0
sip digit timeout: 5
sip blf subscription period: 120
directed call pickup: 1
sip intercom type: 2
sip intercom prefix code: *98
sip intercom line: 1
sip intercom mute mic: 1
sip allow auto answer: 1
prgkey1 type: directory
prgkey2 type: callers
prgkey3 type: speeddial
prgkey3 value: *85
prgkey4 type: dnd
prgkey5 type: speeddial
prgkey5 label: FWD On
prgkey5 value: *72
prgkey6 type: speeddial
prgkey6 label: FWD Off
prgkey6 value: *73
...and here is a .cfg I
...and here is a .cfg I did the majority of the setup for the phones in the aastra.cfg except for the registration info and any customer per-phone settings.
These config files work. You should also check aastra's site to be sure you have their latest firmware.
sip line1 auth name: 107-hop
sip line1 password: xxxxxxx
sip line1 user name: 107-hop
sip line1 display name: 107-hop
sip line1 screen name: Student 2 - 107
# aastra phone options
softkey1 type: speeddial
softkey1 label: "Ext Pickup"
softkey1 value: **
softkey1 states: idle
softkey2 type: icom
softkey2 label: Intercom
softkey2 states: idle
directory 1: hopDirectory.csv
That second file was
That second file was supposed to be a mac-address.cfg file. The forum ate part of my post.
I have always been a fan of the Polycom 550's and 650's. There is also the 560 (550 with gigabit) and a rumored soon-to-be-released 670 (color screen plus gigabit). I cant speak for aastra but polycom phones use a common set of bootrom and software for all their phone models making provisioning a unified process.
I am sure aastra phones provision ok in pbx manager, i have heard of others using them.
have you connected to a specific extension with a soft phone to at least isolate the problem as either being a SIP connectivity problem versus a config retrieval/provisioning issue?