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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031223/7425c3d1/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 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031224/2cc43877/attachment.htm