- osmocom-analog - lists.osmocom.org

Future Directions in Nursing Education and Paper Writing Services

by taylorswiftmerch146＠gmail.com

As nursing education continues to evolve, driven by advancements in healthcare delivery and technological innovation, the role of nursing paper writing services will likely expand to meet the changing needs of students and educators. Future directions in nursing education and paper writing [url=https://www.writinkservices.com/nurs-fpx-4060-assessment-1-health-promo… fpx 4060 assessment 1 health promotion plan ps[/url] services may include: Integration of artificial intelligence and machine learning technologies to enhance the efficiency and effectiveness of paper writing services.Collaboration between academic institutions and paper writing services [url=https://www.writinkservices.com/nurs-fpx-6103-assessment-2-applying-the… fpx 6103 assessment 2 applying the tripartite model[/url] to promote academic integrity and ethical writing practices.Expansion of services to support emerging areas of nursing practice, such as telehealth, informatics, and population health.Conclusion: Navigating the Future of Nursing EducationIn conclusion, the evolution of nursing education reflects a commitment to excellence, innovation, and continuous improvement in preparing nurses to meet the complex healthcare needs of society. Nursing paper writing services play a vital role in supporting this mission by providing students with the resources and support they need to succeed academically. As we look to the future, nursing educators, students, and paper writing [url=https://www.writinkservices.com/nurs-fpx-4030-picot-questions-and-evide… fpx 4030 picot questions and evidence based approach[/url] services must continue to collaborate and innovate to ensure that nursing education remains responsive to the evolving needs of healthcare delivery and patient care.

2 months

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osmocom-analog installation problem

by gvideki79＠gmail.com

HELLO Everybody! I have a problem during installation of osmocom-analog to my computer, i am running Oracle VM virtualbox under Win11 OS. There is a syntax error while running ./configure - line 5619: sytax error near unexpected token 'ALSA' line 5619: PKG_CHECK_MODULES(ALSA, ALSA >= 1,0, with alsa=yes, with alsa=no) "make" and "make install " commands do not work. Please help me if you can, I will send you some screenshots. We appreciate your work! Greetings from HUNGARY! George

2 months, 4 weeks

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Osmocom discourse as alternative to mailing lists

by Harald Welte

Dear Osmocom community, while many people with a long history in FOSS development have no issues at all with mailing lists as primary form of engaging with their community, they have undoubtedly fallen out of fashion in favor of various chat/messaging systems or web based forums. In Osmocom, we've just launched an installation of the discourse forum software available at https://discourse.osmocom.org/ providing an alternative to our traditional mailing lists at https://lists.osmocom.org/ We're looking forward to see whether this web-based approach will facilitate more and/or other people to engage with the Osmocom developer/contributor community. Feel free to join and get the discussions started. If there's a need for more categories or sub-categories, just let one of the moderators know and we can help with that. The old mailing lists will continue to remain available for those who prefer them. -- - Harald Welte <laforge(a)osmocom.org> https://laforge.gnumonks.org/ ============================================================================ "Privacy in residential applications is a desirable marketing option." (ETSI EN 300 175-7 Ch. A6)

9 months

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osmocom-analog to osmo-cc-sip-endpoint local server help

by admin＠ready100.com

osmocom-analog is working. osmo-cc-sip-endpoint is installed and working (i think) however, i'm not sure what my sip-endpoint command line should be... Everything is running on the same machine, I just want to connect my AMPS device to my local asterisk server. I have setup a user for the pbx using the phone number baked into the phone. I have a non standard port (ie not default) for my asterisk server. When I try to use command lines I've seen on the lists I just get: sip.c:1783 error : Failed to create SIP stack object I assume it's just because I have incomplete parameters. Can anyone help? I'm so close!

10 months, 4 weeks

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Is there any good cheap full duplex sdrs for amps?

by stubbertville＠hotmail.com

Howdy! Im new to all this and I already have two AMPS cell phones, but I cant seem to find a cheap sdr I could use with them, the hack rf and lime sdr are either waaaay too expensive or waaaay too rare, any suggestions or help would be welcomed and appreciated!

11 months, 3 weeks

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Re: SIP connector for osmocom-analog?

by Famicoman

Hi all, thanks for all your help so far! I believe I have successfully built sofia-sip and osmo-cc-sip-endpoint. I did have an issue building osmo-cc-sip-endpoint as it was complaining about using AC_CONFIG_MACRO_DIR more than once in configure.ac when running autoreconf. I saw that it was defining both AC_CONFIG_MACRO_DIR and AC_CONFIG_MACRO_DIRS so I commented out the latter and everything seems to have proceeded fine afterwards (though with warnings). I'm leaving this note here in case someone else faces this. I'm running Ubuntu 22.04.2 with everything up to date so I'm guessing the build dependencies have become more strict towards this. I am running amps with: $ amps --limesdr-mini --channel 334 - channel 332 -o --fast-math I am running osmo-cc-sip-endpoint with: $ osmo-cc-sip-endpoint --local 192.168.3.225 --remote 192.168.5.228 -R 8927188(a)192.168.5.228 -A 8927188 secretpasswordhere 192.168.5.228 -v 0 * osmo-cc-sip-endpoint connects to my asterisk server at 192.168.5.228 just fine and it also reports it is connecting to my local ip via the socket amps sets up. The issue I'm facing at the moment is that I'm not getting audio from asterisk coming out of the earpiece on the amps phone. I do hear a bit of white noise and an occasional click (interference I'm guessing) and I know the phone is otherwise good because I used it to call another phone through amps. Based on the asterisk logs, everything looks as it would for any other hardline I have connected but I'm not hearing anything. I'll do some more checks later to make sure asterisk is behaving properly and I'll try to call my amps phone from a hardline and see if there's some sort of one-way audio issue. Does anyone know if osmo-cc-sip-endpoint needs a specific audio codec? I'm only allowing ulaw right now, though I also tried with both ulaw and alaw enabled and noticed no difference. To answer Jordan's question about how I compiled the osmocom-analog-binaries, these are the steps I used: $ autoreconf -if $ ./configure $ make -j4 $ sudo make install I think I'm really close, just not 100% there yet. Cheers, Mike On Wed, Feb 22, 2023 at 3:42 AM <jordan(a)k9fax.us> wrote: > On 2023-02-20 21:42, Famicoman wrote: > > Hello all! I have been enjoying osmocom-analog with my AMPS phones for > > the last few days and was wondering if there has been any work done > > for SIP connectivity with analog phones? I see there is > > 'osmo-sip-connector' software from my searches but as far as I can > > tell, there is no similar software of osmocom-analog. > > > > Thanks, > > Mike > > -- > > osmocom-analog mailing list -- osmocom-analog(a)lists.osmocom.org > > To unsubscribe send an email to osmocom-analog-leave(a)lists.osmocom.org > > That connector does indeed work with Osmocom-analog, it can be a bit > tricky with the lack of documentation however, I have navigated it > before and would be happy to lend a hand. Can you tell me about how you > compiled your osmocom-analog binaries (what options and such used)? >

1 year, 1 month

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Re: SIP connector for osmocom-analog?

by jordan＠k9fax.us

On 2023-02-23 03:34, Famicoman wrote: > Hi all, thanks for all your help so far! > > I believe I have successfully built sofia-sip and > osmo-cc-sip-endpoint. I did have an issue building > osmo-cc-sip-endpoint as it was complaining about using > AC_CONFIG_MACRO_DIR more than once in configure.ac [1] when running > autoreconf. I saw that it was defining both AC_CONFIG_MACRO_DIR and > AC_CONFIG_MACRO_DIRS so I commented out the latter and everything > seems to have proceeded fine afterwards (though with warnings). I'm > leaving this note here in case someone else faces this. I'm running > Ubuntu 22.04.2 with everything up to date so I'm guessing the build > dependencies have become more strict towards this. > > I am running amps with: > $ amps --limesdr-mini --channel 334 - channel 332 -o --fast-math > > I am running osmo-cc-sip-endpoint with: > $ osmo-cc-sip-endpoint --local 192.168.3.225 --remote 192.168.5.228 -R > 8927188(a)192.168.5.228 -A 8927188 secretpasswordhere 192.168.5.228 -v 0 > * > > osmo-cc-sip-endpoint connects to my asterisk server at 192.168.5.228 > just fine and it also reports it is connecting to my local ip via the > socket amps sets up. The issue I'm facing at the moment is that I'm > not getting audio from asterisk coming out of the earpiece on the amps > phone. I do hear a bit of white noise and an occasional click > (interference I'm guessing) and I know the phone is otherwise good > because I used it to call another phone through amps. Based on the > asterisk logs, everything looks as it would for any other hardline I > have connected but I'm not hearing anything. > > I'll do some more checks later to make sure asterisk is behaving > properly and I'll try to call my amps phone from a hardline and see if > there's some sort of one-way audio issue. > > Does anyone know if osmo-cc-sip-endpoint needs a specific audio codec? > I'm only allowing ulaw right now, though I also tried with both ulaw > and alaw enabled and noticed no difference. > > To answer Jordan's question about how I compiled the > osmocom-analog-binaries, these are the steps I used: > $ autoreconf -if > $ ./configure > $ make -j4 > $ sudo make install > > I think I'm really close, just not 100% there yet. > > Cheers, > Mike > > On Wed, Feb 22, 2023 at 3:42 AM <jordan(a)k9fax.us> wrote: > >> On 2023-02-20 21:42, Famicoman wrote: >>> Hello all! I have been enjoying osmocom-analog with my AMPS phones >> for >>> the last few days and was wondering if there has been any work >> done >>> for SIP connectivity with analog phones? I see there is >>> 'osmo-sip-connector' software from my searches but as far as I can >>> tell, there is no similar software of osmocom-analog. >>> >>> Thanks, >>> Mike >>> -- >>> osmocom-analog mailing list -- osmocom-analog(a)lists.osmocom.org >>> To unsubscribe send an email to >> osmocom-analog-leave(a)lists.osmocom.org >> >> That connector does indeed work with Osmocom-analog, it can be a bit >> >> tricky with the lack of documentation however, I have navigated it >> before and would be happy to lend a hand. Can you tell me about how >> you >> compiled your osmocom-analog binaries (what options and such used)? > > > Links: > ------ > [1] http://configure.ac Glad to hear you are getting there! What codec is your asterisk server set to use? You may need to play with the allowed codecs on that end. I remember running into that issue before as well however I do not remember how I went about resolving it. I will be able to play around with my setup again later this weekend and I can let you know how I had mine configured. You may want to ensure under your Asterisk SIP Settings tab that you have ulaw, alaw, and gsm codecs enabled/allowed to start with. You may also be able to specify the codec in the sip connector parameters. The last thing I can think to mention is to ensure your SIP traffic is flowing freely to both ends, SIP through firewalls can be a huge pain in the rear end.

1 year, 2 months

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SIP connector for osmocom-analog?

by Famicoman

Hello all! I have been enjoying osmocom-analog with my AMPS phones for the last few days and was wondering if there has been any work done for SIP connectivity with analog phones? I see there is 'osmo-sip-connector' software from my searches but as far as I can tell, there is no similar software of osmocom-analog. Thanks, Mike

1 year, 2 months

3
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No Device for Transceiver

by carsten.gallini＠gmail.com

https://pasteboard.co/dkAOhAMS7qOh.png https://pasteboard.co/V7C90XoOSC5n.png UHD is able to find the device and set the name so I do have control over the device, I'm not sure what the deal is.

1 year, 4 months

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Paging problem with osmo-cc

by Dave

Hi All, I have an issue where paging doesn't work when using osmo-cc-endpoint with tacs and a USRP1 I can place calls from the MS out over osmo-cc-endpoint via my pbx to the pstn with no problems, if I try to call back in via sip it always fails to page the MS Running the -x option I can call between two MS stations with no issues, paging works and the calls connect Here is an example if I run the command I can make calls from the mobile device and hear the music, I can also press d and call the MS back with no problem, paging appears to work 9/10 times ./tacs --sdr-uhd --channel 323 --channel 318 --channel 320 --samplerate 581818 --sdr-tx-antenna TX/RX --sdr-rx-antenna RX2 2342123456 amps.c: 932 info : (chan 323) Call to mobile station, paging station id '2342123456' amps.c:1186 info : (chan 323) Paging the phone frame.c:3693 info : (chan 323) RX Level: 79% Quality: 80% Polarity: POSITIVE amps.c: 832 info : (chan 323) Paging reply 2342123456 (ESN = removed, Class 3 / Continuous / 20 MHz, TIA/EIA-553 or IS-54A mobile station) amps.c:1170 info : (chan 323) Assigning channel to call to mobile station here is the output with -o and an inbound call via sip endpoint.c: 931 info : Handle message CC-ATTACH-REQ at state IDLE (callref 2) message.c: 516 info : IE_CALLING_INTERFACE name='sip' message.c: 588 info : IE_SOCKET_ADDRESS address='127.0.0.1:4201' endpoint.c: 390 info : Remote peer with socket address '127.0.0.1' and port '4201' and interface 'sip' attached to us. endpoint.c: 394 info : Changing message to CC-ATTACH-CNF. endpoint.c: 931 info : Handle message CC-ATTACH-RSP at state ATTACH-SENT (callref 1) message.c: 516 info : IE_CALLING_INTERFACE name='tacs' message.c: 588 info : IE_SOCKET_ADDRESS address='127.0.0.1:4200' endpoint.c: 308 info : Attached to remote peer "127.0.0.1:4201". UUUUUendpoint.c: 931 info : Handle message CC-SETUP-REQ at state IDLE (callref 3) message.c: 522 info : IE_CALLING_NETWORK type=4(sip) id='' message.c: 516 info : IE_CALLING_INTERFACE name='sip' message.c: 582 info : IE_SDP payload=v=0\no=root 2465 2465 IN IP4 x.x.x.x \ns=session\nc=IN IP4 x.x.x.x \nt=0 0\nm=audio 20644 RTP/AVP 0 8 101\na=rtpmap:0 PCMU/8000\na=rtpmap:8 PCMA/8000\na=rtpmap:101 telephone-event/8000\na=fmtp:101 0-16\na=silenceSupp:off - - - -\na=ptime:20\na=sendrecv\n message.c: 498 info : IE_CALLING type=0(unknown) plan=1(telephony), presentation=0(allowed), screening=3(network provided), number='55555555555' message.c: 510 info : IE_CALLING_NAME name='"55555555555"' message.c: 468 info : IE_CALLED type=0(unknown) plan=1(telephony) number='2342123456' message.c: 492 info : IE_COMPLETE message.c: 516 info : IE_CALLING_INTERFACE name='sip' call.c: 744 info : Received OSMO-CC call from fixed network '55555555555' to mobile '2342123456' call.c: 468 info : Indicate OSMO-CC answer towards fixed network endpoint.c: 931 info : Handle message CC-SETUP-CNF at state INIT-OUT (callref 3) message.c: 498 info : IE_CALLING type=4(subscriber) plan=1(telephony), presentation=0(allowed), screening=3(network provided), number='2342123456' message.c: 582 info : IE_SDP payload=v=0\no=- 3851253277 3851253277 IN IP4 127.0.0.1\ns=session\nc=IN IP4 127.0.0.1\nt=0 0\nm=audio 16384 RTP/AVP 8\na=rtpmap:8 PCMA/8000\n call.c: 751 info : Outgoing call from '55555555555' to '2342123456' amps.c: 932 info : (chan 323) Call to mobile station, paging station id '2342123456' transaction.c: 245 info : Transaction state IDLE -> PAGE transaction.c: 150 info : Created transaction for subscriber '2342123456' endpoint.c: 931 info : Handle message CC-SETUP-COMP-REQ at state CONNECTING-OUT (callref 3) amps.c:1186 info : (chan 323) Paging the phone transaction.c: 245 info : Transaction state PAGE -> PAGE SEND frame.c:2721 info : Transmit: Word 1 - Abbreviated Address Word frame.c:2721 info : Transmit: Word 2 - Extended Address Word (SCC == 11) frame.c:2721 info : Transmit: Word 1 - Abbreviated Address Word frame.c:2721 info : Transmit: Word 2 - Extended Address Word (SCC == 11) frame.c:2721 info : Transmit: Word 1 - Abbreviated Address Word frame.c:2721 info : Transmit: Word 2 - Extended Address Word (SCC == 11) transaction.c: 245 info : Transaction state PAGE SEND -> PAGE REPLY amps.c:1084 notice : (chan 323) Paging timeout, retrying transaction.c: 245 info : Transaction state PAGE REPLY -> PAGE amps.c:1186 info : (chan 323) Paging the phone transaction.c: 245 info : Transaction state PAGE -> PAGE SEND frame.c:2721 info : Transmit: Word 1 - Abbreviated Address Word frame.c:2721 info : Transmit: Word 2 - Extended Address Word (SCC == 11) frame.c:2721 info : Transmit: Word 1 - Abbreviated Address Word frame.c:2721 info : Transmit: Word 2 - Extended Address Word (SCC == 11) frame.c:2721 info : Transmit: Word 1 - Abbreviated Address Word frame.c:2721 info : Transmit: Word 2 - Extended Address Word (SCC == 11) transaction.c: 245 info : Transaction state PAGE SEND -> PAGE REPLY amps.c:1081 notice : (chan 323) Paging timeout, destroying transaction transaction.c: 245 info : Transaction state PAGE REPLY -> CALL RELEASE call.c: 566 info : Call has been released with cause=27 endpoint.c: 931 info : Handle message CC-DISC-REQ at state ACTIVE (callref 3) message.c: 564 info : IE_CAUSE location=10(beyond interworking) isdn_cause=16(normal call clearing) sip_cause=486 socket_cause=0(unset) call.c: 685 info : Received OSMO-CC disconnect from fixed network with cause 16 call.c: 686 info : Call disconnected, releasing! call.c: 495 info : Indicated OSMO-CC release towards fixed network endpoint.c: 931 info : Handle message CC-REL-IND at state DISCONNECTING-OUT (callref 3) message.c: 564 info : IE_CAUSE location=1(private network serving local user) isdn_cause=16(normal call clearing) sip_cause=468 socket_cause=0(unset) frame.c:3693 info : (chan 323) RX Level: 234% Quality: 29% Polarity: POSITIVE amps.c:1068 notice : (chan 323) Release timeout, destroying transaction transaction.c: 163 info : Destroying transaction for subscriber '2342123456' Is this likely to the sdr, I can use multiple boards for tx/rx but I can't figure out how to expose the different cards in uhd as the list command doesn't seem to work for antenna Any help appreciated. Thanks Dave

2 years, 2 months

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Unable to compile with SoapySDR

by jordan＠k9fax.us

Hello, I am having a problem getting osmo analog to compile with soapySDR support. I have ensured all required libraries are installed. My hardware is a Raspberry pi 4 running Ubuntu Server 21.04 64-bit. Here is the error output: Making all in libsdr make[2]: Entering directory '/home/ubuntu/osmocom-analog/src/libsdr' CC soapy.o soapy.c: In function ‘soapy_open’: soapy.c:253:52: warning: passing argument 2 of ‘SoapySDRDevice_setupStream’ makes pointer from integer without a cast [-Wint-conversion] 253 | if (!(rxStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_RX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^~~~~~~~~~~~ | | | int In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:295:22: note: expected ‘SoapySDRStream **’ but argument is of type ‘int’ 295 | SoapySDRStream **stream, | ~~~~~~~~~~~~~~~~~^~~~~~ soapy.c:253:66: warning: passing argument 3 of ‘SoapySDRDevice_setupStream’ makes integer from pointer without a cast [-Wint-conversion] 253 | if (!(rxStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_RX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^~~~~~~~~~~~~~ | | | char * In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:296:15: note: expected ‘int’ but argument is of type ‘char *’ 296 | const int direction, | ~~~~~~~~~~^~~~~~~~~ soapy.c:253:82: warning: passing argument 4 of ‘SoapySDRDevice_setupStream’ from incompatible pointer type [-Wincompatible-pointer-types] 253 | if (!(rxStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_RX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^~~~~~~~ | | | size_t * {aka long unsigned int *} In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:297:17: note: expected ‘const char *’ but argument is of type ‘size_t *’ {aka ‘long unsigned int *’} 297 | const char *format, | ~~~~~~~~~~~~^~~~~~ soapy.c:253:92: warning: passing argument 5 of ‘SoapySDRDevice_setupStream’ makes pointer from integer without a cast [-Wint-conversion] 253 | if (!(rxStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_RX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^ | | | int In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:298:19: note: expected ‘const size_t *’ {aka ‘const long unsigned int *’} but argument is of type ‘int’ 298 | const size_t *channels, | ~~~~~~~~~~~~~~^~~~~~~~ soapy.c:253:95: warning: passing argument 6 of ‘SoapySDRDevice_setupStream’ makes integer from pointer without a cast [-Wint-conversion] 253 | if (!(rxStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_RX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^~~~~~~~~~~~ | | | SoapySDRKwargs * In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:299:18: note: expected ‘size_t’ {aka ‘const long unsigned int’} but argument is of type ‘SoapySDRKwargs *’ 299 | const size_t numChans, | ~~~~~~~~~~~~~^~~~~~~~ soapy.c:253:20: error: too few arguments to function ‘SoapySDRDevice_setupStream’ 253 | if (!(rxStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_RX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^~~~~~~~~~~~~~~~~~~~~~~~~~ In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:294:19: note: declared here 294 | SOAPY_SDR_API int SoapySDRDevice_setupStream(SoapySDRDevice *device, | ^~~~~~~~~~~~~~~~~~~~~~~~~~ soapy.c:378:66: warning: passing argument 3 of ‘SoapySDRDevice_setupStream’ makes integer from pointer without a cast [-Wint-conversion] 378 | if (!(txStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_TX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^~~~~~~~~~~~~~ | | | char * In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:296:15: note: expected ‘int’ but argument is of type ‘char *’ 296 | const int direction, | ~~~~~~~~~~^~~~~~~~~ soapy.c:378:82: warning: passing argument 4 of ‘SoapySDRDevice_setupStream’ from incompatible pointer type [-Wincompatible-pointer-types] 378 | if (!(txStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_TX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^~~~~~~~ | | | size_t * {aka long unsigned int *} In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:297:17: note: expected ‘const char *’ but argument is of type ‘size_t *’ {aka ‘long unsigned int *’} 297 | const char *format, | ~~~~~~~~~~~~^~~~~~ soapy.c:378:92: warning: passing argument 5 of ‘SoapySDRDevice_setupStream’ makes pointer from integer without a cast [-Wint-conversion] 378 | if (!(txStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_TX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^ | | | int In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:298:19: note: expected ‘const size_t *’ {aka ‘const long unsigned int *’} but argument is of type ‘int’ 298 | const size_t *channels, | ~~~~~~~~~~~~~~^~~~~~~~ soapy.c:378:95: warning: passing argument 6 of ‘SoapySDRDevice_setupStream’ makes integer from pointer without a cast [-Wint-conversion] 378 | if (!(txStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_TX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^~~~~~~~~~~~ | | | SoapySDRKwargs * In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:299:18: note: expected ‘size_t’ {aka ‘const long unsigned int’} but argument is of type ‘SoapySDRKwargs *’ 299 | const size_t numChans, | ~~~~~~~~~~~~~^~~~~~~~ soapy.c:378:20: error: too few arguments to function ‘SoapySDRDevice_setupStream’ 378 | if (!(txStream = SoapySDRDevice_setupStream(sdr, SOAPY_SDR_TX, SOAPY_SDR_CF32, &channel, 1, &stream_args))) | ^~~~~~~~~~~~~~~~~~~~~~~~~~ In file included from soapy.c:26: /usr/include/SoapySDR/Device.h:294:19: note: declared here 294 | SOAPY_SDR_API int SoapySDRDevice_setupStream(SoapySDRDevice *device, | ^~~~~~~~~~~~~~~~~~~~~~~~~~ make[2]: *** [Makefile:359: soapy.o] Error 1 make[2]: Leaving directory '/home/ubuntu/osmocom-analog/src/libsdr' make[1]: *** [Makefile:360: all-recursive] Error 1 make[1]: Leaving directory '/home/ubuntu/osmocom-analog/src' make: *** [Makefile:365: all-recursive] Error 1 Thanks for any suggestions and help!

2 years, 9 months

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Hello to all! let's make NMT call to public over SIP

by atmega＠wp.pl

Hey there ! I am excited that this topic started ;-) I am 40 yo guy from Poznan, Poland, about 2 decades ago I had pleasure to work with NMT Centertel Dealer. It brings for me great memories. Recently I managed to collect some NMT phones - at the moment I have more than 10 fully functional NMT phones: Nokia 250, 350, 440, 450, 540, 550, 650, Benefon Delta, Benefon Sigma, Benefon Spica, Motorola Associate 2000. I have four Baofeng radios, buils Linux based pc, bought 2 USB cards and managed to compile, run and work Osmocom - Analog and called phone to phone. Also Computer to phone over headphones. I was also trying to launch LCR, bought ISDN card Fritz, and was attempting to use FritzBox with built - in VOIP - Isdn gate and try to make my NMT phone call PSTN over Voip and mISDN/LCR router. So far no succes. I am not a specialist - self learned electronic - hobbyst. If I could afford it I would buy LimeSDR but for the moment it is a bit expensive for me. Anyway - I have tried OSMO-CC-ENDPOINT as I realised that I will not need any LCR / mISDN / asterisk anymore. I did try many times but all I can get is an update about authentication error with my SIP provider. here is log : osmocom-analog connects to osmo-cc-sip-endpoint, but it looks like SIP-Endpoint does not connects to my SIP provider. I did forward 5060 ports to my linux pc, what could I do wrong: sip.c:1430 debug : Event 29 from SIP stack received (handle=0x55cb861d7980) sip.c: 921 info : Received REGISTER response: 401 Unauthorized (registration) sip.c: 943 debug : Register challenge received sip.c: 178 debug : challenge order received sip.c: 194 notice : No authentication header found sip.c: 954 debug : Register failed, starting register timer sip.c: 956 debug : destroying nua_handle 0x55cb861d7980 (register) using this command line: osmo-cc-sip-endpoint -r mysipnumber@mysipserverip -l localIPosmocom-analog -A SipAccountUser SipAccountPass --remote-auth --register SIPAccountUser@SipServerIp --cc 'screen-called-in "" "myNMTphone"' -P myPublicIP -v 0 Any ideas what do I do wrong? Many greetings for All people in osmocom-analog subjects. I am fascinated about this historical technology. Would be very happy if I could call somebody from my NMT phone ! Regards for all people who work with osmocom-analog subject. Respect to all! Jacek

3 years

3
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Osmo-CC documentation

by Andreas Eversberg

Hi, Osmo-CC endpoints are still work in progress. I have seen that people are already started to play with it, so I started a documentation for all endpoints, which is also work in progress. This document will help to set up different endpoints. One example is interfacing an endpoint (osmocom-analog) with SIP. See attachment. The osmo-cc-alsa-endpoint that is also described in the doc is not yet available on git.osmocom.org. Temporarily you can download it here: http://download.eversberg.eu/temp/osmo-cc-alsa-endpoint.tar.gz (@Harald: Can you add this repository?) Also it would be nice to have Osmo-CC as project page at: http://osmocom.org/projects/ There I can upload the latest documentation. Regards, Andreas

3 years

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Odp: Re: Osmo-CC documentation

by atmega＠wp.pl

I wonder ... is there any chance to personalize - replace announcement tones in NMT system... like unacaliable message, instead of current music... regards Jacek Dnia 20 marca 2021 18:34 Andreas Eversberg <andreas(a)eversberg.eu> napisał(a): Attached the new version od Osmo-CC documentation. On 20.03.21 11:59, Andreas Eversberg wrote: Hi, Osmo-CC endpoints are still work in progress. I have seen that people are already started to play with it, so I started a documentation for all endpoints, which is also work in progress. This document will help to set up different endpoints. One example is interfacing an endpoint (osmocom-analog) with SIP. See attachment. The osmo-cc-alsa-endpoint that is also described in the doc is not yet available on git.osmocom.org. Temporarily you can download it here: download.eversberg.eu download.eversberg.eu (@Harald: Can you add this repository?) Also it would be nice to have Osmo-CC as project page at: osmocom.org osmocom.org There I can upload the latest documentation. Regards, Andreas -- osmocom-analog mailing list osmocom-analog(a)lists.osmocom.o lists.osmocom.org lists.osmocom.org

3 years, 1 month

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test, please ignore

by Harald Welte

test if the mailing list works as expected -- - Harald Welte <laforge(a)gnumonks.org> http://laforge.gnumonks.org/ ============================================================================ "Privacy in residential applications is a desirable marketing option." (ETSI EN 300 175-7 Ch. A6)

3 years, 1 month

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osmocom-analog