Hi all,
I am new to the use of the Osmocom project, it is indeed a very nice job.
I am currently trying to set up a configuration with a Asterisk PBX server and I have 2 questions:
1/ RTP configuration
The SIP part (sip-connector vs Asterisk connection) works well so far, the communication starts but with no audio.
I noticed that the RTP flux is sent to localhost instead of my server address (set as remote in sip-connector.cfg) and I was wondering if there is any possibility to send the RTP flow to an address which is not localhost ?
sip
local 0.0.0.0 5069
remote 127.0.0.1 5060
2/ codec issue
In a configuration where all the Osmocom servers (MSC, MGW, BSC…) and Asterisk are on the same machine, it got a message from my asterisk server, saying that no codec can be found to start a communication. By default, the wiki/manuals states that gsm has to be used but perhaps I am missing something in the BSC configuration, especially in the codec choice.
<--- SIP read from UDP:10.184.10.162:5069 --->
INVITE sip:899@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.184.10.162:5069;rport;branch=z9hG4bKpe1UXZU1amUja
Max-Forwards: 70
From: <sip:422@0.0.0.0:5069>;tag=vyQKX32r72ZyQ
To: <sip:899@127.0.0.1:5060>
Call-ID: cd65c5a0-fdbf-1238-51a9-000c29cfd753
CSeq: 949096397 INVITE
Contact: <sip:10.184.10.162:5069>
User-Agent: sofia-sip/1.12.11devel
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Length: 133
v=0
o=Osmocom 0 0 IN IP4 127.0.0.1
s=GSM Call
c=IN IP4 127.0.0.1
t=0 0
m=audio 4016 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=sendrecv
<------------->
--- (13 headers 8 lines) ---
Sending to 10.184.10.162:5069 (no NAT)
Sending to 10.184.10.162:5069 (no NAT)
Using INVITE request as basis request - cd65c5a0-fdbf-1238-51a9-000c29cfd753
No matching peer for '422' from '10.184.10.162:5069'
== Using SIP RTP CoS mark 5
Got SDP version 0 and unique parts [Osmocom 0 IN IP4 127.0.0.1]
Found RTP audio format 3
Found audio description format GSM for ID 3
[2020-04-20 17:38:19] NOTICE[14918][C-00000015]: chan_sip.c:10957 process_sdp: No compatible codecs, not accepting this offer!
Thanks for your help
Laurent
Hello Mr. Welte,
I am writing to ask you please, if it is possible to provide me with documents on the GSUP Client program. I have to implement my own application and send the required GSUP messages to the HLR to get subscriber data from hlr in Osmocom.
Many thanks for your considerations.
Reza
i am trying to read the sim card using sim reader but not able to read
i used command pcsc_scan but its taking huge time for scanning and only
scanning
waiting for the first reader... like this please check whats the issue
Hello,
apologies for the generic question, but how can I contribute to the
pysim project?
I made a couple of fixes and added some stuff I needed for some tests,
but looks like github rejects the pull requests, project is not listed in
gerrit, there is no specific mail list, and the osmocom git repo rejects
any push :-(
Kind regards
Hi,
we are looking for some help creating open source mobile communication
infrastructure in the context of a search and rescue (SAR) mission at sea.
We would like to use the osmocom stack as sensing and communication
infrastructure at sea.
How precise is the timing advance for localization?
Neels mentioned that "modern" mobile stations equipped with a GPS receiver can
report the location to the operator. Could you point me to the right standard?
Is there any reliable and open source tooling to determine the direction of the
signal with multiple receivers (e.g. SDRs) in close proximity?
You can find out more about the project here:
https://www.hs-augsburg.de/searchwing/
Sorry for the mostly German content.
Thanks for your help!
Best Regards
Philipp Borgers
Hi,
I am setting up a nitb/nano3g right now and have the basics working with the nano3g talking to hnbgw and services running on my nitb box (rpi 3-something).
I have two of the same phones: ZTE Obsidian with Android 5.1 which are able to connect to my network.
When I try to call the other phone I don't get connected. It takes about 1 minute for the call to fail.
Once the phone mentioned some problem with "access configuration for normal calls".
Attached are my configs and logs from nano3g trace log and hnbgw journal during a failed call.
Any help would be appreciated. I am sure I did something obviously wrong.
I'll try and read the configs carefully to make sure that matches with what is up on the wiki.
Thanks,
Craig
Dear Developers,
I am trying to build the osmo-euse-demo on the Osmocom VM, executing the following commands:
#Install required development libraries:
sudo apt-get install libosmo-gsup-client-dev
#Get the source code and a missing header file
wget https://raw.githubusercontent.com/osmocom/osmo-hlr/master/src/osmo-euse-dem…
wget https://raw.githubusercontent.com/osmocom/osmo-hlr/master/include/osmocom/h…
mkdir -p osmocom/hlr/
mv logging.h osmocom/hlr/
#Compile source
gcc -I. -o osmo-euse-demo osmo-euse-demo.c -losmocore -losmo-gsup-client -losmogsm -ltalloc
#Run
./osmo-euse-demo
But i am now facing this error :
E: Could not get lock /var/lib/dpkg/lock-frontend - open (11: Resource temporarily unavailable)
E: Unable to acquire the dpkg frontend lock (/var/lib/dpkg/lock-frontend), is another process using it?
Can you please tell me what is the reasin behind this error !? And how can i solve it ?
Many thanks for your consideraion,
Reza
Hello dears!
I try to connect 2 STPs to each other via Sigtran M3UA protocol.
But I don't understand, why my connection is fault. Could you please check
my configuration?
STP1 STP2
point-code 0.12.4 point-code 0.12.4
port 2905 port 2906
IP 172.18.141.1 IP 172.18.142.2
Me configuration is:
STP1
cs7 instance 1
network-indicator reserved
point-code 0.12.4
asp stp2 2906 2905 m3ua
local-ip 172.18.141.1
remote-ip 172.18.142.2
role asp
sctp-role client
as stp2 m3ua
asp stp2
traffic-mode override
routing-key 0 0.25.0
route-table system
STP2
cs7 instance 1
network-indicator reserved
point-code 0.25.0
asp stp1 2905 2906 m3ua
local-ip 172.18.142.2
remote-ip 172.18.141.1
role sg
as stp1 m3ua
asp stp1
traffic-mode override
routing-key 0 0.12.4
route-table system
listen m3ua 2906
accept-asp-connections dynamic-permitted
Thanks in advance!
Hello all,
1) When an SMS is received from SMPP, the validity_minutes is not set (value 0).
The consequence is that email is deleted from database too early in case of failure.
In file: src/libmsc/smpp_openbsc.c function submit_to_sms translate convert from submit_sm_t to gsm_sms.
But some field of gsm_sms are not filled, in particularly validity_minutes.
I think that validity_minutes must be computed from submit_sm_t.schedule_delivery_time or submit_sm_t.validity_period.
Sorry I don’t know exactly content of these values.
2) In file src/libmsc/sms_queue.c function sub_ready_for_sm,
before sending SMS using the function gsm411_send_sms,
the SMS is not place in pending list. Like it is done in functions sms_submit_pending and sms_send_next.
The consequence is that when handset do a Location Updating Request,
only one SMS is send if they are many SMS for this subscribers.
Because in function sms_sms_cb call of
pending = sms_find_pending(network->sms_queue, sig_sms->sms->id);
will always return 0.
I try the following fix and was working
but I don’t know if it possible that an SMS was already in the list?
+ struct gsm_sms_queue *smsq = net->sms_queue;
+ struct gsm_sms_pending *pendingSms = sms_pending_from(smsq, sms);
+ if (!pendingSms) {
+ LOGP(DLSMS, LOGL_ERROR,
+ "Failed to create pending SMS entry.\n");
+ sms_free(sms);
+ return 0;
+ }
+ llist_add_tail(&pendingSms->entry, &smsq->pending_sms);
gsm411_send_sms(net, vsub, sms);
return 0;
}
Thanks you,
Denis