[asterisk-ss7] Combined linksets and mtp3d
Michael Mueller
ss7box at gmail.com
Thu Jan 12 17:26:45 CST 2012
Better chance of working without doing additional development. ssp to stp1
would be primary. ssp to stp2 would be alternative.
A1------------stp1----------ssp
| | |
| +-----+ |
| | |
+---------------+ |
| | |
+------+ | |
| | |
A2------------stp2-----------+
On Thu, Jan 12, 2012 at 9:50 AM, German Becker <german.becker at gmail.com>wrote:
> HI Michael, thanks for the reply. By hosts I mean asterisk boxes, that
> would make them SSPs Both of them are connected with one signalling link to
> one STP (different STP for each host) wich in turn will connect to the
> destination STP(s). I'll try to sketch the scenario
>
> -------------
> | Asterisk1 |--voice trunks-
> | mtp3d |----------------|STP 1|----------|SSP| (destination point
> code)
> ------------- /
> | (tcp) /
> ------------ /
> |Asterisk2 |--voice trunks /
> |mtp3d |----------------|STP 2|--
> ------------
>
> As for the variant. it is ITU. And weather it would be primary/alternate
> or "load balanced" is under discussion with the connected party. However
> load bancing is a possibility.
>
> What I would like to achieve is for incomming messages, both mtp3d route
> them to the corresponding host (according to cic and ord opc); and for
> outgoing messages, each asterisk deceide on which link (or linkset) send
> it.
>
> I thought something like these could be achieved, based on this part of
> the below(l4isuop.c), However, I don't get how should i configure it.
>
> switch (link->linkset->loadshare) {
> case LOADSHARE_NONE:
> if (!link->schannel.mask)
> slink = link;
> break;
> case LOADSHARE_LINKSET:
> if (linkset->n_slinks)
> slink = linkset->slinks[cic % linkset->n_slinks];
> break;
> case LOADSHARE_COMBINED_LINKSET:
> {
> int n_slinks = 0;
> int six;
> for (lsi = 0; lsi < n_linksets; lsi++)
> if (linksets[lsi].enabled)
> if (&linksets[lsi] == linkset ||
> (is_combined_linkset(linkset,
> &linksets[lsi])))
> n_slinks += linksets[lsi].n_slinks;
> if (n_slinks) {
> six = cic % n_slinks;
> n_slinks = 0;
> for (lsi = 0; lsi < n_linksets; lsi++)
> if (linksets[lsi].enabled)
> if (&linksets[lsi] == linkset ||
> (is_combined_linkset(linkset,
> &linksets[lsi]))) {
> if (six - n_slinks <
> linksets[lsi].n_slinks) {
> slink = linksets[lsi].slinks[six -
> n_slinks];
> break;
> }
> n_slinks += linksets[lsi].n_slinks;
> }
> }
> }
> break;
> }
>
>
>
> On Thu, Jan 12, 2012 at 11:29, Michael Mueller <ss7box at gmail.com> wrote:
>
>> are those 2 "hosts" STPs or SSPs? what you describe could be a combined
>> linkset if connected to STPs and using ANSI SS7; if using ITU then this
>> might be a primary/alternate linkset scenario; my experience says the
>> combined linkset concept exists in ANSI SS7 and not in ITU
>>
>> On Thu, Jan 12, 2012 at 8:15 AM, German Becker <german.becker at gmail.com>wrote:
>>
>>> Hi,
>>>
>>> I have a scenario with 2 hosts, 2 links on each host, one of the links
>>> with signaling on each host and the DPC for all the voice TS reachable
>>> through both signaling links.
>>>
>>> I'm not sure if I need to create a single linkset with two signaling
>>> links on it (one per host), or two combined linksets.
>>> I tried the 2 links aproach, but when one of the links is down (i.e.
>>> asterisk lose connection with the remote mtp3d), all the TS are set to
>>> block, instead of singnaling through the other link.
>>>
>>> Has anyone set up a similar scenario?
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-ss7 mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-ss7 mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20120112/c1b4896b/attachment.htm>
More information about the asterisk-ss7
mailing list