Hi All, i dont 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031222/ad2acc89/attachment.htm
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@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
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@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Balaji NJL Sent: 23 December 2003 23:04 To: asterisk-users@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@yahoo.com> To: asterisk-users@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 <http://us.rd.yahoo.com/evt=21486/*http:/f1.pg.photos.yahoo.com/ph/spsim plenol?.file=ny_ts_splash.html> ________________________________ Do you Yahoo!? Yahoo! Photos - Get your photo on the big screen in Times Square <http://us.rd.yahoo.com/evt=21486/*http://f1.pg.photos.yahoo.com/ph//sps implenol?.file=ny_ts_splash.html> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031223/0cf9fc9b/attachment.htm
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@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@lists.digium.com  
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Balaji NJL
Sent: 23 December 2003 23:04
To:   asterisk-users@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@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
---------------------------------
Do you Yahoo!?
Yahoo! Photos - Get your photo on the big screen in Times Square
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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@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@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@lists.digium.com    
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Balaji NJL
Sent: 23 December 2003     23:04
To:     asterisk-users@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@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  
  
---------------------------------
  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
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