A ramble about MSC, sip-connector and asterisk.

[Sandi Suhendro]

Osmocom stacks doesnt have any problem with some pbx especially asterisk.
If you facing cannot get call progress via pstn, please give us details log.

One i can remain you, nat also need to be configure to yes for many case.

regards,
Sandi / DUO

On Sat, Jan 12, 2019, 19:48 Gullik Webjorn <gullik.webjorn at corevalue.se
wrote:

> Hi,
>
> Some weeks ago I managed to integrate osmobts with my asterisk server.
> In the beginning, I did
>
> not get any call progress, ( as opposed to running WITHOUT asterisk ). I
> realized that defining
>
> the msc to sipconnector socket effectively disabled all the logic
> related to this.
>
> To remedy the situation, I set an asterisk "progress statement" whenever
> the extensions.conf
>
> indicated a number belonging to my GSM network. This works OK, but now,
> I don't get call progress
>
> on calls toward PSTN. These extensions do not have "progress"
> statements, since all SIP phones
>
> interpret the call progress sip messages such as RINGING.
>
>
> I am interested in your comments whether msc + sipconnector should
> emulate mobiles as "sip phones",
>
> it does seem to just silently drop call progress messages, and is this
> the way it should work?
>
> The functionality IS there, since when osmobts was bridging the calls,
> call progress WAS there.
>
> Well, I kludged up a solution for the GSM-only situation, and this seems
> suboptimal.
>
> I guess some clever asterisk hacking would solve that, possibly having a
> redundant progress statement
>
> on ALL extensions would do it, but would I not lose "actual" call
> progress signals from the PSTN,
>
> since asterisk would emulate them? Another way would be a small "local"
> asterisk.....before the *real* one.
>
> Boldly moving on to MultiBTS and handover....
>
> Gullik
>
>
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