please check the following details in your asterisk configuration and on
your phones. These are the settings that work for me:
sip.conf
------------
[general]
limitonpeers=yes
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
useclientcode=yes
canreinvite=yes
[user1]
secret=user1
host=dynamic
username=user1
callerid="user1 <97>"
dtmfmode=rfc2833
context=local
type=friend
callgroup=1
pickupgroup=1
qualify=yes
vmexten=80297
call-limit=20
subscribecontext=local
extensions.conf
--------------------
exten => 97,hint,SIP/user1
exten => 98,hint,SIP/smguenther
On the SNOM phones:
Support broken Registrar: ON
Use user:phone: OFF
Filter Packets from Registrar: OFF
Function Key P6:
ACTIVE / EXTENSION / <sip:98@192.168.0.101>
Hope that helps,
Stefan
--
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in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Geschaeftsfuehrer
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
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Schulungen Installationen
Beratung Support
Voice-over-IP-Loesungen
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