[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 other softphones. the only phone thats working with GS is Xtern. MSN and SJ doesnt work. Is this a known issue.
 
Thanks,
-B
 
  ----- Original Message ----- 
  From:   Balaji NJL   
  To: asterisk-users at lists.digium.com   
  Sent: Tuesday, December 23, 2003 7:05   PM
  Subject: Re: [Asterisk-Users] MSN to GS -   Call drops in 10 secs
  

  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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