[Asterisk-Users] MSN to GS - Call drops in 10 secs
Craig Waddington
craig at xmbsystems.com
Tue Dec 23 17:34:14 MST 2003
Balaji,
I also have the same issue. Works fine on any phone except GS for me.
After a bit of research I found a post saying set the phone to "offer
only one codec set".
It looks like we have to set the phone to use one codec - GSM
I am concerned that you cant use passwords when logging in to * using
Messenger.
Craig.
________________________________
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Balaji NJL
Sent: 23 December 2003 23:04
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] MSN to GS - Call drops in 10 secs
resending.
Can anyone help me in trying to understand what would be the problem.
appreciate ur time. i need to get this working.
thanks a lot,
-B
----- Original Message -----
From: Balaji NJL <mailto:bajjeen at yahoo.com>
To: asterisk-users at lists.digium.com
Sent: Monday, December 22, 2003 8:15 PM
Subject: [Asterisk-Users] MSN to GS - Call drops in 10 secs
Hi All,
i dont know what changes i made recently but i am unable to hold
the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS
and vice versa works fine. i am not behind NAT.Also MSN to MSN works
fine too.
my SIP details
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
;context = default
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
;My SIP phone - GS
[2000]
type=friend
username=2000
secret=qweqwe
host=dynamic
context=from-sip
mailbox=2000
dtmfmode=inband
;MSN Msgr
[2002]
type=friend
host=dynamic
insecure=yes
dtmfmode=inband
;dtmfmode=rfc2833
context=from-sip
mailbox=2002
;auth=plaintext
i did a SIP trace
it says Format=UKN
CSeq=BYE
thanks for the help,
-Balaji
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