[Asterisk-Users] MSN to GS - Call drops in 10 secs

[Balaji NJL]

v\:* {	BEHAVIOR: url(#default#VML)}o\:* {	BEHAVIOR: url(#default#VML)}w\:* {	BEHAVIOR: url(#default#VML)}shape {	BEHAVIOR: url(#default#VML)}st1\:*{behavior:url(#default#ieooui) }     i tried with only GSM too. With only GSM it doesnt even connect to GS. Then someone recommended to use ulaw and alaw and that helped. But the call drops after 10 secs. i did a 'sip debug' and what i found is that MSN doesnt even  recognize that call is in progress and then drops the call. Any way i can increase this or disable this option.
 
thanks,
-B
  ----- Original Message ----- 
  From:   Craig   Waddington 
  To: asterisk-users at lists.digium.com   
  Sent: Tuesday, December 23, 2003 4:34   PM
  Subject: RE: [Asterisk-Users] MSN to GS -   Call drops in 10 secs
  

    
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     

        
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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