osmocom-analog April 2023

osmocom-analog@lists.osmocom.org

3 discussions

SIP connector for osmocom-analog?
by Famicoman 15 Apr '26

15 Apr '26

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

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No Device for Transceiver
by carsten.gallini＠gmail.com 08 Apr '26

08 Apr '26

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.

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Re: SIP connector for osmocom-analog?
by jordan＠k9fax.us 08 Apr '26

08 Apr '26

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 April 2023