[Asterisk-Users] MSN to GS - Call drops in 10 secs
Balaji NJL
bajjeen at yahoo.com
Tue Dec 23 16:03:39 MST 2003
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
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
---------------------------------
Do you Yahoo!?
Yahoo! Photos - Get your photo on the big screen in Times Square
---------------------------------
Do you Yahoo!?
Yahoo! Photos - Get your photo on the big screen in Times Square
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031223/ccc7f42e/attachment.htm
More information about the asterisk-users
mailing list