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)?
>
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.
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
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
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!
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
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
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.orgosmocom.org There I can upload the latest documentation.
Regards,
Andreas
--
osmocom-analog mailing list
osmocom-analog(a)lists.osmocom.o
lists.osmocom.orglists.osmocom.org
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)
