Hello,
Yes, I also believe that this is some sort of codec issue. Here is my sip.conf
file:
[201]<?xml:namespace prefix = o ns =
"urn:schemas-microsoft-com:office:office" />
type=friend
;secret=201
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=no
host=dynamic
dtmfmode=rfc2833
dial=SIP/201
context=from-internal
canreinvite=no
callerid=device <201>
[202]
type=friend
;secret=202
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=no
host=dynamic
dtmfmode=rfc2833
dial=SIP/202
context=from-internal
canreinvite=no
callerid=device <202>
Note: The "secret" is commented out so that there is no authentication
when registering with the Jain-Sip phones.
Thanks,
-----Original Message-----
From: Gerald A [mailto:geraldablists at gmail.com]
Sent: Tuesday, September 11, 2007 5:12 PM
To: Kutman DK at ADM(Mat) DAEPM(R&CS)@Ottawa-Hull
Subject: Re: [asterisk-users] Chan_sip Entry
Hi again,
On 9/11/07, Kutman.DK at forces.gc.ca < Kutman.DK at forces.gc.ca
<mailto:Kutman.DK at forces.gc.ca> > wrote:
I am trying to get to Jain Sip softphones to call one another via an Asterisk
server. When I call from phone 1 to phone 2 there is audio transmission both
ways, but when I call from phone 2 to phone 1 I don't get audio transmission
and reception both ways. When I look at the asterisk log file it has an entry
which says:
"Oooh, format changed to 2".
Usually this is a codec selection problem. Are both Jain's the same version?
Maybe posting your sip.conf for the phones might help.
Thanks,
Gerald.
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