Dear Osmocom community,
It's my pleasure to announce the next OsmoDevCall at
July 9, 2021 at 20:00 CEST
at
https://meeting4.franken.de/b/har-xbc-bsx-wvs
This meeting will have the following schedule:
20:00 meet + greet
20:15 hands-on tutorial by miaoski: Setting up open5gs
21:00 USSE: unstructured supplementary social event [*]
22:00 close of call
Attendance is free of charge and open to anyone with an interest
in Osmocom.
More information about OsmoDevCall, including the schedule
for further upcoming events can be found at
https://osmocom.org/projects/osmo-dev-con/wiki/OsmoDevCall
Looking forward to meeting you on Friday.
Best regards,
Harald
[*] this is how we started to call the "unstructured" part of osmocom
developer conferences in the past, basically where anyone can talk about
anything, no formal schedule or structure.
--
- Harald Welte <laforge(a)osmocom.org> http://laforge.gnumonks.org/
============================================================================
"Privacy in residential applications is a desirable marketing option."
(ETSI EN 300 175-7 Ch. A6)
Hi all.
I'm testing to see if inter-BSC handover will work between RBS and
osmo-bts, given that support for handover from an E1 to RTP call is not
yet supported.
Among all kinds of other things going on, I'm not finding it easy to
understand how to configure everything for inter bsc handover.
I know there are two videos (at least) from Neels at OsmoDevCon, on the
subject of handover and then inter-MSC handover.
I have one DUG/RUS configurarion with the e1d, bsc/msc/mgw/hlr running
on the associated PC and one remote SysmoBTS, which is running osmo-bsc
and it's associated MGW locally. All that is fine, I can call between
the two BTSs successfully.
I'm guessing I should add the neighbours manually in each bsc but I'm
not sure what to configure in the MSC.
Maybe I'll figure it out this evening, but I thought it best to also
write most given the time zones and most of you guys will see this
before I get back to work tomorrow.
Thanks!
k
Hello All,
Sorry about creating noise on this list, but I could not find the correct place to raise this: people.osmocom.org seems to be down. I have checked it from multiple internet connections and also with DownDetector and the result was always the same.
I noticed this when I tried to watch a recording of the OsmoDevCall about SS7 and SIGTRAN - the link to the video timed out.
Apologies again for asking for a sysadmin task on this list. In case there is anything I could help with or there is a mirror of these videos I don’t know of (yet) feel free to tell me about it.
Thank you!
Domi
Hi Devs,
I have successfully compiled and integrated osmoSTP with our product. I
have a question regarding to assotiaction establishment:
Currently we'll initiate the associtation establishment from our product
and STP will reply to our INIT message. Is it possible to start the
establishment from STP side? I haven't found any cli
command/configuration for this.
Thanks,
Laszlo
Dear Osmocom community,
I am using pysim to read values of ISIM. However, it just returns "None, None". And the aids of my SIM card is ['a0000000871002ff86ffff89ffffffff']. It seems does not have ISIM application on my SIM card.
How should I read values of ISIM?
Best regards,
Zishuai CHENG
Dear Osmocom community,
It's my pleasure to announce the next OsmoDevCall at
June 11, 2021 at 20:00 CEST
at
https://meeting4.franken.de/b/har-xbc-bsx-wvs
This meeting will have the following schedule:
20:00 meet + greet
20:15 presentation by keith: "Screen Sharing peek at TIC A.C. infrastructure in Oaxaca"
21:00 USSE: unstructured supplementary social event [*]
22:00 close of call
TIC A.C. is an operator of Osmocom based community cellular networks in
indigenous communities of the Mexican state of Oaxaca. Keith works with
Rhizomatica and TIC A.C. and will give us some live insight into how
they operate
Attendance is free of charge and open to anyone with an interest
in Osmocom.
More information about OsmoDevCall, including the schedule
for further upcoming events can be found at
https://osmocom.org/projects/osmo-dev-con/wiki/OsmoDevCall
Looking forward to meeting you on Friday.
Best regards,
Harald
[*] this is how we started to call the "unstructured" part of osmocom
developer conferences in the past, basically where anyone can talk about
anything, no formal schedule or structure.
--
- Harald Welte <laforge(a)osmocom.org> http://laforge.gnumonks.org/
============================================================================
"Privacy in residential applications is a desirable marketing option."
(ETSI EN 300 175-7 Ch. A6)
Hi all,
I'm using pySim-Shell with a SysmoISIM-SJA2 as well as a SysmoUSIM-SJS1 and
on both cards I get quite a few "File not found" errors trying to access
different EFs. I'll use EF.NCP-IP as an example to illustrate the issue I'm
having below. My initial steps when I plug in the card are as follows:
1) ./pySim-shell -p0
2) verify_adm <adm1 key>
3) select ADF.USIM
Now when I run "dir" in ADF.USIM it lists a number of EFs I'm interested
in, including EF.NCP-IP. Since this requires service number 80 to be
available in EF.UST, I run "select EF.UST" followed by
"ust_service_activate 80". This runs without issue.
When I run select ADF.USIM now followed by select EF.NCP-IP, the card
returns a "File not found" error. So my questions are:
Is this file not supported on the SJA2 and SJS1?
Is there a way for me to add those files to ADF.USIM?
Is there a list of the known supported EFs of the SJA2 and SJS1? I looked
and couldn't find anything.
Thank you in advance!
Hi all,
I feel like I may be missing something about how SIM cards work here, but I’m unable to figure out how to load the ISIM application on my SysmoISIM-SJA2s, so I’m reaching out here for some help.
I started with pySim-shell and listed the directories under the MF directory and it shows only ADF.USIM, so I decided to use shadysim_isim.py to list out the applications on the card and was shown this:
AID: a0000000620001, State: 01, Privs: 00
AID: 4a6176656c696e2e6a637265, State: 01, Privs: 00
AID: a0000000620101, State: 01, Privs: 00
AID: a0000000620102, State: 01, Privs: 00
AID: a0000000620201, State: 01, Privs: 00
AID: a000000062020801, State: 01, Privs: 00
AID: a00000006202080101, State: 01, Privs: 00
AID: a0000000620002, State: 01, Privs: 00
AID: a0000000620003, State: 01, Privs: 00
AID: a000000062010101, State: 01, Privs: 00
AID: a00000015100, State: 01, Privs: 00
AID: a0000000090005ffffffff8911000000, State: 01, Privs: 00
AID: a0000000090005ffffffff8912000000, State: 01, Privs: 00
AID: a0000000090005ffffffff8913000000, State: 01, Privs: 00
AID: a0000000090005ffffffff8911010000, State: 01, Privs: 00
AID: a0000000871005ffffffff8913100000, State: 01, Privs: 00
AID: a0000000871005ffffffff8913200000, State: 01, Privs: 00
AID: a0000000090003ffffffff8910710001, State: 01, Privs: 00
AID: a0000000090003ffffffff8910710002, State: 01, Privs: 00
AID: a0000000090005ffffffff8915000000, State: 01, Privs: 00
AID: a00000015141434c, State: 01, Privs: 00
Instance AID: a00000015141434c00
The SysmoUSIM/ISIM manual pointed me to Annex E of 3GPP 101.220 which says the prefix of the ISIM application is A00..00871004.
So my questions are:
Am I missing something and the app is installed?
If not, is there a way to load and install the ISIM application?
Thanks in advance everyone.
Dear List
I am trying to use latest Osmo-core suite (osmo:stp,hlr,msc,bsc,bts,mgw)
with limesdr mini but facing issue as soon power rampup starts.
is there any stable release which I can use with limesdr mini
--
Akib Sayyed
Matrix-Shell
akibsayyed(a)gmail.com
akibsayyed(a)matrixshell.com
Mob:- +91-966-514-2243
Hi all.
I would like to implement some functionality that existed as a kind of
happy accident?, in the osmocom-nitb
That is - that media endpoints in INVITEs/200s and re-INVITEs from a SIP
UA to osmo-sip-connector just ended up being transparently communicated
to osmo-bts. This made it really easy to take the pbx out of the audio
stream.
I wanted to share where I am at with that.
freeswitch even has a console command for this: "uuid_media" can switch
the pbx in and out of the stream during the call. In the case of a MS to
MS call on the same bts, we just connect the two osmo-bts media
endpoints to each other. Freeswitch can schedule to playback message on
calls and such like - for "your credit ran out" message for example -
and it works pretty good. FS knows to put itself back into the stream
with a re-INVITE with it's own IP in the SDP before playing the audio
and invites itself back out of the stream as soon as playback is finished.
I've been looking on and off, not that I've given it a huge amount of
time, at how to replicate this with the BSC/MSC/MGW/SIP combination.
First thing was to fix up the BSS messages and Global Call Reference
generation. (TS 23.284) Given that Max had done most of this, that's
pretty easy, although I got lost at one point down a rabbit hole of
trying to figure out more than I needed to with the FSM and BSS
messages, and that put me off working on it for a while.
Anyway, https://gerrit.osmocom.org/c/osmo-msc/+/24236 exists and
"works" - that is you can do LCLS with either mgw-loop or bts-loop
configured on the BSC - but only for the internal mncc, obviously as the
osmo-sip-connector still does not know what to do with the GCR (global
call reference)
I thought about various hackish things to do. (maintain a table of
call-ids and their gcr on the sip conn and add an X-Osmo-CallID header
to the SIP) but I've also been looking at the specs.
TS 29.164 (section 6) says that the GCR should be encapsulated in a
binary encoded ISUP payload, there are pointers in the spec to ITU
Q.1912.5 (Section5 and thereabouts) So the ISUP messages should be added
to SIP messages, requiring a multipart MIME attachment to the INVITE
with the SDP and the ISUP. Also see RFC3204
What I don't know is how sip UEs will respond to these. I wonder will
they decode them, I doubt it, some searching for SIP-I support in
FreeSwitch confirms no support.
So I'm thinking that I need to serialise the GCR and add it to an "X-"
SIP header, so we at get it back on the B-leg.
So yes, I need to just do this, but I got stalled a bit again with my
lack on programming skills trying to learn how to serialise the GCR the
"right" way. - but once I figure that out, then I can at least test what
happens and see how the MSC and MGW behaves with the MNCC messages that
result from SIP re-INVITES.
Maybe that just works, but I don't think so, I think I will need to
implement more BSS LCLS messages to actually change the LCLS states, not
just update media endpoints.
Sometimes I wonder here what should happen in the sip conn and what in
the MSC.
Anyway, It's been on my mind to write this up quickly to the ML just in
case somebody has a "OH... better not do it like that" type comment..
Thanks
k.
