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@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@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@lists.osmocom.org To unsubscribe send an email to osmocom-analog-leave@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)?
Hi Mike,
you need to specify your RTP peer, if the remote RTP endpoint is on a different machine. The default is 127.0.0.1, which the remote SIP endpoint @192.168.5.228 reach. As 192.168.3.225 is your local network interface to talk to the other SIP endpoint, you should add --cc "rtp-peer 192.168.3.225":
amps --limesdr-mini --channel 334 - channel 332 -o --fast-math --cc "rtp-peer 192.168.3.225"
Note that if you define STUN server or public IP at osmo-cc-sip-endpoint, the "rtp-peer" gets automatically translated into public IP address. But in your case there seems to be no NAT in between.
I use 'sngrep' to check that by looking at the INVITE message.
Regards,
Andreas
On 2/23/23 04:34, Famicoman wrote:
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@192.168.5.228 -A 8927188 secretpasswordhere 192.168.5.228 -v 0 *
Thanks everyone for your help! I think I have it working at this point. Adding --cc "rtp-peer 192.168.3.225" did the trick and I could call into my pbx and listen to a recording. I still need to make sure incoming calls work and ring the amps phone (which I'll attempt this week) but I have no reason to believe it wouldn't work as expected.
Andreas noted correctly that there is no NAT in between my machine and the asterisk server. If we were to suppose that there was a NAT and the asterisk server lived somewhere on the Internet, would I need to take any other considerations aside from using my external IP address in the osmo-cc-sip-endpoint arguments in place of the internal IP? I'm just thinking about a future scenario if I want to take the phone somewhere else for testing.
Thanks, Mike.
On Sun, Feb 26, 2023 at 2:54 AM Andreas Eversberg andreas@eversberg.eu wrote:
Hi Mike,
you need to specify your RTP peer, if the remote RTP endpoint is on a different machine. The default is 127.0.0.1, which the remote SIP endpoint @192.168.5.228 reach. As 192.168.3.225 is your local network interface to talk to the other SIP endpoint, you should add --cc "rtp-peer 192.168.3.225":
amps --limesdr-mini --channel 334 - channel 332 -o --fast-math --cc "rtp-peer 192.168.3.225"
Note that if you define STUN server or public IP at osmo-cc-sip-endpoint, the "rtp-peer" gets automatically translated into public IP address. But in your case there seems to be no NAT in between.
I use 'sngrep' to check that by looking at the INVITE message.
Regards,
Andreas
On 2/23/23 04:34, Famicoman wrote:
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@192.168.5.228 -A 8927188 secretpasswordhere 192.168.5.228 -v 0 *
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Andreas Eversberg -
Famicoman