- osmocom-analog - lists.osmocom.org

Having trouble transmitting with USRP
by h416984j＠gmail.com 27 Nov '24

27 Nov '24

Hi, I have been interested in trying out Osmocom-analog's AMPS implementation. I purchased an Ettus B210 clone device and built osmocom-analog from source as described on osmocom-analog.eversberg.eu. Everything went well until I started the program and tried to transmit, I received no Tx (output) and the built-in graphs and visualizers had strange output unlike what was expected: https://imgur.com/a/Ghr1cju. I used Ubuntu 22.04 and UHD driver 4.7.0. I am truly at a loss at what is causing the strange output, I also tried launching with soapy and several other arguments. Additionally, the LimeSDR Mini V2 I had previously acted the same way as well, so I'm unsure what the problem is. Any advice will be appreciated, and more information can be provided. Thank you. Full log with debug enabled shared below: sudo amps --sdr-uhd --channel 334 --channel 332 --fast-math --sdr-rx-gain -12 --sdr-tx-gain 6 --samplerate 999999 -v 1 DOPTIONS INFO options.c:128 Config file '/root/.osmocom/analog/amps.conf' seems not to exist, using command line options only. DOPTIONS INFO options.c:296 Command line option '--sdr-uhd' DOPTIONS INFO options.c:294 Command line option '--channel', parameter '334' DOPTIONS INFO options.c:294 Command line option '--channel', parameter '332' DOPTIONS INFO options.c:296 Command line option '--fast-math' DOPTIONS INFO options.c:294 Command line option '--sdr-rx-gain', parameter '-12' DOPTIONS INFO options.c:294 Command line option '--sdr-tx-gain', parameter '6' DOPTIONS INFO options.c:294 Command line option '--samplerate', parameter '999999' DOPTIONS INFO options.c:280 Command line option '-v' ('--verbose'), parameter '1' Selected System ID (SID) 40 belongs to: Salinas, CA (GTE Mobile) San Francisco, CA (GTE Mobile) San Jose, CA (GTE Mobile) San Rosa, CA (GTE Mobile) Santa Barbara, CA (GTE Mobile) DDSP DEBUG dsp.c:150 Generating sine table for SAT signal. DAMPS DEBUG amps.c:597 Creating 'AMPS' instance for channel = 334 of band B (sample rate 999999). DSENDER DEBUG sender.c:77 (chan 334) Creating 'Sender' instance DJITTER INFO jitter.c:129 (334) Reset jitter buffer. DJITTER INFO jitter.c:113 (334) Created jitter buffer. (samperate=8000, target_window=60ms, max_window=1000ms, flag:latency=true flag:repeat=true) DJITTER INFO jitter.c:129 (334) Reset jitter buffer. DJITTER INFO jitter.c:113 (334) Created jitter buffer. (samperate=999999, target_window=60ms, max_window=1000ms, flag:latency=true flag:repeat=true) DDSP DEBUG dsp.c:202 (chan 334) Init DSP for transceiver. DSENDER DEBUG sender.c:308 (chan 334) Maximum deviation: 8.0 kHz, Maximum modulation: 10.0 kHz DSENDER DEBUG sender.c:309 (chan 334) Deviation at speech level: 2.9 kHz DDSP DEBUG dsp.c:218 Use 99.9999 samples for full bit duration @ 999999. DDSP DEBUG dsp.c:234 Bit window length: 100 DDSP DEBUG dsp.c:235 -> Samples in window to analyse level left of edge: 25..49 DDSP DEBUG dsp.c:236 -> Samples in window to analyse level right of edge: 50..74 DDSP DEBUG dsp.c:175 Generating smooth ramp table. DDSP DEBUG dsp.c:259 Sat detection interval is 66 ms. DDSP DEBUG dsp.c:927 (chan 334) SAT detector reset: Initial state. DAMPS DEBUG amps.c:399 (chan 334) State change: (NULL) -> IDLE DAMPS INFO amps.c:685 (chan 334) Entering IDLE state, sending Overhead/Filler frames on combined control & paging channel. DDSP DEBUG dsp.c:927 (chan 334) SAT detector reset: Change to FOCC. DDSP INFO dsp.c:944 (chan 334) Change mode to FOCC DDSP DEBUG dsp.c:967 (chan 334) Reset FSK frame transmitter, due to setting DSP mode. DAMPS NOTICE amps.c:638 Created channel #334 (System B) of type 'CC/PC' = combined control & paging channel Base station on channel 334 ready (combined control & paging channel), please tune transmitter to 880.020 MHz and receiver to 835.020 MHz. (-45.000 MHz offset) DAMPS NOTICE amps.c:558 Channel number 332 belongs to a control channel, but your channel type 'voice channel' requires it to be on a voice channel number. Some phones may reject this, but all my phones don't. DAMPS DEBUG amps.c:597 Creating 'AMPS' instance for channel = 332 of band A (sample rate 999999). DSENDER DEBUG sender.c:77 (chan 332) Creating 'Sender' instance DJITTER INFO jitter.c:129 (332) Reset jitter buffer. DJITTER INFO jitter.c:113 (332) Created jitter buffer. (samperate=8000, target_window=60ms, max_window=1000ms, flag:latency=true flag:repeat=true) DJITTER INFO jitter.c:129 (332) Reset jitter buffer. DJITTER INFO jitter.c:113 (332) Created jitter buffer. (samperate=999999, target_window=60ms, max_window=1000ms, flag:latency=true flag:repeat=true) DDSP DEBUG dsp.c:202 (chan 332) Init DSP for transceiver. DSENDER DEBUG sender.c:308 (chan 332) Maximum deviation: 8.0 kHz, Maximum modulation: 10.0 kHz DSENDER DEBUG sender.c:309 (chan 332) Deviation at speech level: 2.9 kHz DDSP DEBUG dsp.c:218 Use 99.9999 samples for full bit duration @ 999999. DDSP DEBUG dsp.c:234 Bit window length: 100 DDSP DEBUG dsp.c:235 -> Samples in window to analyse level left of edge: 25..49 DDSP DEBUG dsp.c:236 -> Samples in window to analyse level right of edge: 50..74 DDSP DEBUG dsp.c:175 Generating smooth ramp table. DDSP DEBUG dsp.c:259 Sat detection interval is 66 ms. DDSP DEBUG dsp.c:927 (chan 332) SAT detector reset: Initial state. DAMPS DEBUG amps.c:399 (chan 332) State change: (NULL) -> IDLE DAMPS INFO amps.c:692 (chan 332) Entering IDLE state (sending silence / no RF) on voice channel. DDSP DEBUG dsp.c:927 (chan 332) SAT detector reset: Disable FVC. DDSP INFO dsp.c:962 (chan 332) Change mode from FVC to OFF DDSP DEBUG dsp.c:967 (chan 332) Reset FSK frame transmitter, due to setting DSP mode. DAMPS NOTICE amps.c:638 Created channel #332 (System A) of type 'VC' = voice channel Base station on channel 332 ready (voice channel), please tune transmitter to 879.960 MHz and receiver to 834.960 MHz. (-45.000 MHz offset) DSDR DEBUG sdr.c:149 Open SDR device DSDR INFO sdr.c:168 Require bandwidth of each channel is 2 * (8.0 deviation + 10.0 modulation) = 36.0 KHz DSDR INFO sdr.c:345 We move center frequency between the two channels in the middle, to prevent them from overlapping with DC level. DSDR INFO sdr.c:133 TX Spectrum: 2 1 ---------------------------------------+--------------------------------------- DSDR INFO sdr.c:135 Frequency 1 = 880.0200 MHz DSDR INFO sdr.c:135 Frequency 2 = 879.9600 MHz DSDR INFO sdr.c:357 Total bandwidth (two sidebands) for all TX Frequencies: 960 00 Hz DSDR INFO sdr.c:365 Using center frequency: TX 879.990000 MHz DSDR DEBUG sdr.c:370 Frequency #0: TX offset: 0.030000 MHz DSDR DEBUG sdr.c:370 Frequency #1: TX offset: -0.030000 MHz DSDR INFO sdr.c:391 Using gain: TX 6.0 dB DSDR INFO sdr.c:457 We move center frequency between the two channels in the middle, to prevent them from overlapping with DC level. DSDR INFO sdr.c:133 RX Spectrum: 2 1 ---------------------------------------+--------------------------------------- DSDR INFO sdr.c:135 Frequency 1 = 835.0200 MHz DSDR INFO sdr.c:135 Frequency 2 = 834.9600 MHz DSDR INFO sdr.c:469 Total bandwidth (two sidebands) for all RX Frequencies: 96000 Hz DSDR INFO sdr.c:477 Using center frequency: RX 834.990000 MHz DSDR DEBUG sdr.c:482 Frequency #0: RX offset: 0.030000 MHz DSDR DEBUG sdr.c:482 Frequency #1: RX offset: -0.030000 MHz DSDR INFO sdr.c:492 Using gain: RX -12.0 dB DSDR INFO sdr.c:525 Using local oscillator offset: -1000000 Hz DUHD INFO uhd.c:59 Using device args "" DUHD INFO uhd.c:60 Using stream args "" DUHD INFO uhd.c:61 Using tune args "" DUHD INFO uhd.c:64 Creating USRP with args ""... [INFO] [UHD] linux; GNU C++ version 11.4.0; Boost_107400; UHD_4.7.0.0-0ubuntu1~ jammy1 [INFO] [B200] Detected Device: B210 [INFO] [B200] Operating over USB 3. [INFO] [B200] Initialize CODEC control... [INFO] [B200] Initialize Radio control... [INFO] [B200] Performing register loopback test... [INFO] [B200] Register loopback test passed [INFO] [B200] Performing register loopback test... [INFO] [B200] Register loopback test passed [INFO] [B200] Setting master clock rate selection to 'automatic'. [INFO] [B200] Asking for clock rate 16.000000 MHz... [INFO] [B200] Actually got clock rate 16.000000 MHz. [INFO] [B200] Asking for clock rate 31.999968 MHz... [INFO] [B200] Actually got clock rate 31.999968 MHz. [INFO] [B200] Asking for clock rate 31.999968 MHz... [INFO] [B200] OK [INFO] [B200] Asking for clock rate 31.999968 MHz... [INFO] [B200] OK [INFO] [B200] Asking for clock rate 31.999968 MHz... [INFO] [B200] OK DUHD NOTICE uhd.c:422 Given RX gain -12.000 is not supported, we use 0.000 [INFO] [B200] Asking for clock rate 31.999968 MHz... [INFO] [B200] OK [INFO] [B200] Asking for clock rate 31.999968 MHz... [INFO] [B200] OK | | | /|===|\ -=O=- / |\ /| \ ~ | AMPS / | X | \ ~ / |===| \ / | |\ \ / /| | \ \ ~ |--| / / |===| \ \ /|\/|\ / / | | \ \ / |--| \ / / | | \ \ / | |\ \ / / |===| \ \ __ / /|--| \ \ / / | | \ \ \_ / / | | \ \ / / | | \ \ \ / / | | \ \______/ / | | \| \| \========================================================================= \_______|/\|______ _***_*__**_***___**_*_| X | |MMMM| \__ |\/| \______/* ** * * * ** * |/ \|______________|MMMM| _____ \_|/\| * * * * ** * * * * * * |\ /|*____/ ~ ~ ~ ~ / ~ ~\_______I__I_________*______*_*___*__*___*__*_*_| X |/ ~ ~ _________ _______\ _ _ _ _ _ _ _ _ _ _ |/_\/ ~ ______/ _ _ _ _ _ _ _ _/~ ~ ____/ ~ ~ ~ _ _ _ _ _ _ _ /_____/ ~ ____________ __/~\______________ _ _ _ _ _/ ~ ~ __________/ ~ ~ ~ ~ ~ ~ \____________________________/ / ~ ~ ~ ~ ~ DSDR DEBUG sdr.c:707 Create threads! DFRAME INFO frame.c:2723 Transmit: Word 1 - System Parameter Overhead DFRAME DEBUG frame.c:2734 T1T2: 3 = Overhead Message (Type field) DFRAME DEBUG frame.c:2736 DCC: 0 (Digital color code field) DFRAME DEBUG frame.c:2736 SID1: 20 (First part of the system identi fication field) DFRAME DEBUG frame.c:2734 EP: 0 = No (Extended Protocol Capabilit y Indicator) DFRAME DEBUG frame.c:2734 AUTH: 0 = No (Support of authentication p rocedures described in TIA/EIA-136-510) DFRAME DEBUG frame.c:2734 PCI: 0 = No (Set to 1 if Control Channel can assign digital traffic channels) DFRAME DEBUG frame.c:2736 NAWC: 4 (Number of additional words comin g field) DFRAME DEBUG frame.c:2734 OHD: 6 = Word 1 of system parameter messa ge (Overhead Message Type field) DFRAME DEBUG frame.c:2736 P: 1530 (Parity field) DFRAME INFO frame.c:2723 Transmit: Word 2 - System Parameter Overhead DFRAME DEBUG frame.c:2734 T1T2: 3 = Overhead Message (Type field) DFRAME DEBUG frame.c:2736 DCC: 0 (Digital color code field) DFRAME DEBUG frame.c:2734 S: 1 = Yes (Serial number field) DFRAME DEBUG frame.c:2734 E: 1 = Yes (Extended address field) DFRAME DEBUG frame.c:2734 REGH: 1 = Yes (Registration field for hom e stations) DFRAME DEBUG frame.c:2734 REGR: 1 = Yes (Registration field for roa ming stations) DFRAME DEBUG frame.c:2734 DTX Support: 0 = DTX Not Supported (Indicates th e nature of DTX supported on an analog voice) DFRAME DEBUG frame.c:2734 N-1: 20 = 21 (N is the number of paging channels in the system) DFRAME DEBUG frame.c:2734 RCF: 1 = Yes (Read-control-filler field) DFRAME DEBUG frame.c:2734 CPA: 1 = Yes (Combined paging/access fie ld) DFRAME DEBUG frame.c:2734 CMAX-1: 20 = 21 (CMAX is the number of acce ss channels in the system) DFRAME DEBUG frame.c:2734 END: 0 = No (End indication field) DFRAME DEBUG frame.c:2734 OHD: 7 = Word 2 of system parameter messa ge (Overhead Message Type field) DFRAME DEBUG frame.c:2736 P: 437 (Parity field) DFRAME INFO frame.c:2723 Transmit: Registration ID Message DFRAME DEBUG frame.c:2734 T1T2: 3 = Overhead Message (Type field) DFRAME DEBUG frame.c:2736 DCC: 0 (Digital color code field) DFRAME DEBUG frame.c:2736 REGID: 196474 (Registration ID field) DFRAME DEBUG frame.c:2734 END: 0 = No (End indication field) DFRAME DEBUG frame.c:2734 OHD: 0 = Registration ID (Overhead Messa ge Type field) DFRAME DEBUG frame.c:2736 P: 614 (Parity field) DFRAME INFO frame.c:2723 Transmit: Registration Increment Global Action Message DFRAME DEBUG frame.c:2734 T1T2: 3 = Overhead Message (Type field) DFRAME DEBUG frame.c:2736 DCC: 0 (Digital color code field) DFRAME DEBUG frame.c:2734 ACT: 2 = Registration increment (Global action field) DFRAME DEBUG frame.c:2736 REGINCR: 300 (Registration increment field) DFRAME DEBUG frame.c:2736 RSVD: 0 (Reserved for future use) DFRAME DEBUG frame.c:2734 END: 0 = No (End indication field) DFRAME DEBUG frame.c:2734 OHD: 4 = Global action (Overhead Message Type field) DFRAME DEBUG frame.c:2736 P: 448 (Parity field) DFRAME INFO frame.c:2723 Transmit: Access Type Parameters Global Action Message DFRAME DEBUG frame.c:2734 T1T2: 3 = Overhead Message (Type field) DFRAME DEBUG frame.c:2736 DCC: 0 (Digital color code field) DFRAME DEBUG frame.c:2734 ACT: 9 = Access type parameters (Global action field) DFRAME DEBUG frame.c:2734 BIS: 0 = Ignore Idle-Busy after access ( Busy-Idle status field) DFRAME DEBUG frame.c:2734 PCI_HOME: 0 = No (Home Protocol Capability In dicator) DFRAME DEBUG frame.c:2734 PCI_ROAM: 0 = No (Roam Protocol Capability In dicator) DFRAME DEBUG frame.c:2734 BSPC: 0 = Reserved for backward compatibil ity (Base Station Protocol Capability field) DFRAME DEBUG frame.c:2734 BSCAP: 0 = Reserved for backward compatibil ity (Base Station Core Analog Protocol field) DFRAME DEBUG frame.c:2736 RSVD: 0 (Reserved for future use) DFRAME DEBUG frame.c:2734 END: 1 = Yes (End indication field) DFRAME DEBUG frame.c:2734 OHD: 4 = Global action (Overhead Message Type field) DFRAME DEBUG frame.c:2736 P: 1790 (Parity field) DFRAME INFO frame.c:2723 Transmit: Control-Filler Message DFRAME DEBUG frame.c:2734 T1T2: 3 = Overhead Message (Type field) DFRAME DEBUG frame.c:2736 DCC: 0 (Digital color code field) DFRAME DEBUG frame.c:2736 010111: 23 (bit combination 23) DFRAME DEBUG frame.c:2734 CMAC: 4 = -10 dbW (100 Milliwatts) (Contr ol mobile attenuation field) DFRAME DEBUG frame.c:2736 SDCC1: 0 (Supplementary Digital Color Code s) DFRAME DEBUG frame.c:2736 11: 3 (bit combination 3) DFRAME DEBUG frame.c:2736 SDCC2: 0 (Supplementary Digital Color Code s) DFRAME DEBUG frame.c:2736 1: 1 (bit combination 1) DFRAME DEBUG frame.c:2734 WFOM: 1 = Yes (Wait-for-overhead-message field) DFRAME DEBUG frame.c:2736 1111: 15 (bit combination 15) DFRAME DEBUG frame.c:2734 OHD: 1 = Control-Filler (Overhead Messag e Type field) DFRAME DEBUG frame.c:2736 P: 1921 (Parity field) DFRAME INFO frame.c:3515 (chan 334) Subsequent system/filler frames are not sho wn, to prevent flooding the output. DSDR DEBUG sdr.c:952 read delay = 2.040 ms DSDR DEBUG sdr.c:884 write delay = 2.040 ms DSDR DEBUG sdr.c:952 read delay = 2.040 ms DSDR DEBUG sdr.c:884 write delay = 2.040 ms DSDR DEBUG sdr.c:952 read delay = 2.040 ms DSDR DEBUG sdr.c:884 write delay = 2.040 ms DSDR DEBUG sdr.c:952 read delay = 4.080 ms DSDR DEBUG sdr.c:884 write delay = 2.040 ms DSDR DEBUG sdr.c:952 read delay = 2.040 ms DSDR DEBUG sdr.c:884 write delay = 2.040 ms DSDR DEBUG sdr.c:952 read delay = 2.040 ms DSDR DEBUG sdr.c:884 write delay = 2.040 ms DSDR DEBUG sdr.c:952 read delay = 2.040 ms DSDR DEBUG sdr.c:884 write delay = 2.040 ms DSDR DEBUG sdr.c:952 read delay = 2.040 ms DSDR DEBUG sdr.c:884 write delay = 2.040 ms [continues like this]

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Re: Fwd: osmocom-analog Website down?
by Andreas Eversberg 21 Jun '24

21 Jun '24

Please try again. I just fixed some entries in the DNS. It should work with IPv6/IPv4 now. On 6/21/24 06:56, Harald Welte wrote: > > > -------- Original Message -------- > From: Tyler Bufkin <wz5tx(a)outlook.com> > Sent: 21 June 2024 04:04:46 CEST > To: "osmocom-analog-owner(a)lists.osmocom.org" <osmocom-analog-owner(a)lists.osmocom.org> > Subject: Website down? > > Hello, > I was just trying to pull up your website and I can't seem to get it to load. > I am wanting to get my analog cellular project up and going. > Thanks! > > WZ5TX > David “Tyler” Bufkin > 817-846-4693 >

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Problem getting AMPS to start
by Shawn 05 May '24

05 May '24

Hello all. My name is Shawn. I am having a problem getting osmocom amps to successfully start. lsusb shows my B210 on Bus 3, Device 3. uhd is installed. uhd find devices shows: INFO] [UHD] linux; GNU C++ version 11.4.0; Boost_107400; UHD_4.6.0.0-141-g0f2007fc -------------------------------------------------- -- UHD Device 0 -------------------------------------------------- Device Address: serial: 191796 name: MyB210 product: B210 type: b200 I am attempting to start AMPS with: amps --sdr-uhd -k 334 -k 332 -p -d -F no --sysinfo sid=18 1234567890 The result is: DSENDER ERROR sender.c:240 No device for transceiver! There is more text before and after the above message. I can paste it if you like. But I think the problem is I am not using proper syntax to tell AMPS to use the my B210. I have tried SoapySDR and have received the same results. I would appreciate help in how I can properly tell the osmocom software to use my B210. Thank you very much, -Shawn

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Future Directions in Nursing Education and Paper Writing Services
by taylorswiftmerch146＠gmail.com 19 Feb '24

19 Feb '24

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osmocom-analog installation problem
by gvideki79＠gmail.com 26 Jan '24

26 Jan '24

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

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Osmocom discourse as alternative to mailing lists
by Harald Welte 27 Jul '23

27 Jul '23

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)

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osmocom-analog to osmo-cc-sip-endpoint local server help
by admin＠ready100.com 31 May '23

31 May '23

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!

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Is there any good cheap full duplex sdrs for amps?
by stubbertville＠hotmail.com 04 May '23

04 May '23

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!

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Re: SIP connector for osmocom-analog?
by Famicoman 06 Mar '23

06 Mar '23

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)? >

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Re: SIP connector for osmocom-analog?
by jordan＠k9fax.us 24 Feb '23

24 Feb '23

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.

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osmocom-analog