[asterisk-dev] SIP over TCP/SCTP

[Dmitry Andrianov]

JFYI,
http://svn.digium.com/svn/asterisk/team/oej/codename-pineapple

Not Found
The requested URL /svn/asterisk/team/oej/codename-pineapple was not
found on this server.
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Apache Server at svn.digium.com Port 80

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Olle E
Johansson
Sent: Sunday, June 10, 2007 5:08 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] SIP over TCP/SCTP


10 jun 2007 kl. 13.49 skrev Ibrar Ahmed:

> Hi,
>
> I have been working on asterisk since last 2/3 years. I have  
> implemented pay phone channel based on mgcp protocol(chan_mgcp.c)  
> and also worked on ss7 channel implementation. Now I feel I should  
> contribute on community. I am working on sip over TCP/SCTP. I have  
> some question about this
>
> 1 - Is any body working on this.
Yes for TCP, I have never got any requests for SCTP though.

> 2 - How much important this feature is.
Very

See http://www.codename-pineapple.org for more information about  
changes that are being worked on.

/O

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