[asterisk-dev] SIP over TCP/SCTP

Ibrar Ahmed ibrar.ahmad at gmail.com
Mon Jun 11 07:55:16 CDT 2007


Stefan van der Eijk wrote:
> On 6/10/07, Ibrar Ahmed <ibrar.ahmad at gmail.com> wrote:
>> 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.
> It seems that people have been working on it:
> http://bugs.digium.com/view.php?id=4903
>
Yes it looks like some body is working on it, but i don't know whats the 
status of that project.


>> 2 - How much important this feature is.
>> 3 - Is there any feature important than this to implement.
>>
>>   --Ahmed
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-dev mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>



More information about the asterisk-dev mailing list