[Asterisk-Users] Codecs and negotiations

[Mike Meyer]

Stig,

	I have not had that problem. Assuming there are no other CODECs allowed
under the general and user sections, Your sip.conf looks OK. Can the
stanaphone be configured itself for the codecs via a on-board web page
like the GS and snoms? In my case the settings in the sip.conf always
overrode what was defined on the phone itself though. So I don't really
know what is going on with yours.

Mike

Subject: [Asterisk-Users] Codecs and negotiations
To: <asterisk-users at lists.digium.com>
Message-ID: <007001c4a59e$6b37eea0$1402a8c0 at HESS>
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For some reason I now seem unable to control which codec is chosen. The
problem happens with outgoing calls to Stanaphone. Even if I chose
disallow=all and allow=ulaw as the only codecs it connects with GSM.
 
Has anyone else got problems with these settings? Any suggestions? As I
recalled it, such a setup would not establish a call if the ulaw-codec
was not offered by the provider. Stanaphone has assured that their
preferred codec is ulaw....
 
This is what I have in sip.conf:
 
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
externip = frode.dyndns.org
localnet = 192.168.0.0/16
context=stighess
tos=lowdelay
maxexpirey=3600         ; Max length of incoming registration we allow
defaultexpirey=60               ; Default length of incoming/outoing
registratio
bandwidth=high
disallow=all
allow=ulaw
.......
[stanaphone]
type=friend
username=91438xxxx
fromuser=91438xxxxx
secret=*********
host=sip.stanaphone.com
context=stighess
fromdomain=216.128.82.18
insecure=very
nat=yes
canreinvite=no
disallow=all
allow=ulaw

__
Stig Hess