[asterisk-ss7] Where's Matt been? Well, here's the explanation

Attila Domjan adomjan at tvnet.hu
Fri Mar 26 04:33:53 CDT 2010


I have some remark 

- should add many members to asterisk channel structure to carry the SS7
attributes, eliminate the tons of SS7_* channel variables
- implement the new channel structure memebers to pass on other channel
types (SIP, IAX2)
- adding type flag on SIP/IAX2 channels to describe the peer is
"custemer" -> hide callerid if presentation not allowed, don't pass many
additianal channel attributes, "trunk" -> pass many attributes to peer
as the channel driver can
- I had many dirty hacking around dtmf in chan_dahdi for for alarm
systems. I had turn off dtmf detection on dahdi -> SIP/MGCP g711 calls.
The alarm systems communicates 10digits/sec... So would be nice if a new
bridging option, turning off any dtmf detection on uncompressed media
I have not posted all off the patches to bugs.digium.com you can find it
in my svn.

This version is stable, I'm using in my production system, have 10k+
customers, with modified asterisk + modifed chan_mgcp (NCS,
realtime, PacketCable COPS).


On Thu, 2010-03-25 at 23:40 -0500, Matthew Fredrickson wrote:
> Hey all,
> It's been a long time.  I apologize for my quietness for the last while 
> here, it has been a very busy year this last year with some of the other 
> projects I've been working on.
> I just wanted to let everyone know that I'm still alive, and haven't 
> given up or forgotten about libss7, and Asterisk with SS7.
> I appreciate very much Attila's great work on getting libss7 polished 
> up.  He's done a very good job taking it to the next level, and has done 
> a great job helping everyone out.  Thanks very much Attila.
> As far as getting his code merged back in, I was in the midst of this a 
> while ago, but had to stop due to personal time issues as well as some 
> other potentially architectural issues that subsequently came up. 
> Hopefully we can get that moving again in the next little bit though.
> I actually have been quite busy, and hopefully have some things to show 
> for it, two things actually.
> I'll not go into too much detail tonight (as it's getting quite late) 
> but I'm looking for some hungry testers, that wouldn't mind beating up 
> on some probably alpha level code.
> They are:
> 1.) libss7 point code clustering support.
> Basically, you can have Asterisk boxes share signalling links now using 
> this code.  Although the signalling links are physically terminated on 
> other machines, you can plug bearer T1s/E1s into other Asterisk boxes 
> and virtually utilize the signalling links of the other boxes.
> 2.) A new channel driver, called chan_ccs, that allows, among other 
> things, you to control MGCP media gateways for bearer trunks, instead of 
> having to locally terminate them on the asterisk box that's controlling 
> the signalling links.  There is also code in the same branch that has 
> chan_ccs that modified chan_mgcp so that Asterisk can act as a media 
> gateway (since I don't have any good real media gateways to test on). 
> This basically means you can have Asterisk TDM channel scalability up to 
> (in the ideal state) the same level as you can do with SIP with no 
> media, per box.
> In essence, this is turning Asterisk into a "true" softswitch, allowing 
> native bridging between media gateways and any other RTP endpoint 
> (including other media gateways).  This also means that you don't have 
> to terminate bearer T1s/E1s on the main signalling box.
> -- So, what does this mean for you, you may be asking?
> These are really the next steps in making big SS7 work with Asterisk. 
> They both allow for scaling a point code across multiple asterisk 
> machines, and distribution of bearers on different boxes than the ones 
> that contain signalling links.
> Like I said though, most of the work is in a functional but early phase, 
> and so I need some people that are interested enough in the added 
> functionality that they're willing to work with potential hickups along 
> the way.
> Some of the changes I've had to make to libss7 have made it further more 
> difficult to merge Attila's changes back in, which is the other reason 
> why it has been so long and it still has not been merged.
> If you're interested, either reply to me or this thread and let me know.
> Thanks again,
> Matthew Fredrickson
> Digium, Inc.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 190 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100326/2da54fcc/attachment.pgp 

More information about the asterisk-ss7 mailing list