[Asterisk-Users] Quesiton about SIP and MSN

[Thorsten Neumann]

Hi there

I got this to work. Sign-in and registration work fine. Also, a SJphone
SIP registered client making a call to the MSN client works. However, once
the call is established and I start any audio, the CLI spits out:

File codec_gsm.c, Line 132 (gsmtolin_framein): Huh? A GSM frame that isn't
a multiple of 33 bytes long from RTP (65)?

What does this mean? Has this perhaps been fixed in a later CVS update?

Regards
Thorsten Neumann
Rand Afrikaans University, South Africa

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of WipeOut .
> Sent: 07 April 2003 09:35 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Quesiton about SIP and MSN
> 
> Here is a minimal config..
> 
> ---for sip.conf---
> [1235]
> type=friend
> host=dynamic
> ------------------
> 
> ---extensions.conf---
> exten => 1235,1,Dial(SIP/1235)
> exten => 1235,2,Congestion
> ---------------------
> 
> This is a very basic setup and you will have to investigate the various
> options to see what you need for your particular config.. The extension
is
> 1235 which for simplicity is the same as the phone name but you can call
> the phone whatever you like just remember to change it in the
> extensios.conf file.
> 
> For you sip client (MSN) the username will be "1235" (without the
quotes)
> and the password will be blank..
> 
> Hope that helps..
> 
> ----- Original Message -----
> From: "it"
> Date: Mon, 7 Apr 2003 08:31:19 -0700
> To:
> Subject: [Asterisk-Users] Quesiton about SIP and MSN
> 
> 
> Hi,I would like to ask a silly question. I want to test * with MSN
> messager. I want to know if I should setup a SIP server in my LAN.  I
> don't know what the MSN's sign-in name means. And how to configure it in
> sip.conf and extensions.conf.  A example would be appreciated.
> 
> john
> --
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