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
On 07.03.2021 21:09, atmega@wp.pl wrote:
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
Dear all,
please disregard my fist message in this thread - I was being a fast-fingered Freddy.
What I wanted to post is this:
With the help of the great examples on the mailinglist, I am able to successfully make incoming and outgoing C-Net/PSTN calls via Sipgate, a German SIP provider.
As my PBX is SIP-capable (it is an Auerswald Compact 5000R), I wanted to attach osmo-cc-sip-endpoint to an extension of my PBX.
The registration is successful, but when I receive a call, C-Netz (or Zeitansage) registers and accepts the incoming call, but the PBX does not understand the call is connected and keeps ringing the extension.
Using SNGREP, I believe I found the cause.
Following a 183 Session Progress message from osmo-cc-sip-endpoint to the PBX, the PBX transmits PRACK, and osmo-cc-sip endpoint responds with a message 504 Method Not Allowed. The PBX ignores this message 504 and keeps ringing the extension until I disconnect the call.
Sipgate, by contrast, responds by sending ACK, not PRACK, and the call connects successfully.
I'm considering opening a ticket with Auerswald on this and would greatly appreciate your expert views. To my understanding, a 504 Method Not Allowed should prompt the PBX to fall back to ACK, or at least realise there is an issue, and yet it seems to ignore the Method Now Allowed response from osmo-cc-sip-endpoint. Does this appear correct, or might there be an issue with Sofia-SIP?
I'm attaching the capture file in case this were useful.
Thank you so much in advance for your insights.
BR, Markus
On 29.03.21 23:34, Markus Stamm wrote:
Following a 183 Session Progress message from osmo-cc-sip-endpoint to the PBX, the PBX transmits PRACK, and osmo-cc-sip endpoint responds with a message 504 Method Not Allowed. The PBX ignores this message 504 and keeps ringing the extension until I disconnect the call.
Sipgate, by contrast, responds by sending ACK, not PRACK, and the call connects successfully.
Hi Markus,
can you try the attached patch? Just add the line as shown in the patch. It should allow "PRACK" messages and make the stack reply to them correctly.
Regards,
Andreas
Hi Andreas:
can you try the attached patch? Just add the line as shown in the patch. It should allow "PRACK" messages and make the stack reply to them correctly.
Thank you for this - the signalling should now work correctly, as the call connects immediately once the internal extension 221 (against which osmo-cc-sip-endpoint registers) is called. There is one issue remaining - once connected, no audio is transmitted. I don't believe this is related to this patch, though, as my Sipgate account continues working normally with the patch applied. As I am testing this in an openSUSE VM, I suspect it will be more of a firewall issue - or perhaps yet another Auerswald issue. I will be looking into this in more detail a little later today. I'm also attaching a SNGREP capture just in case this were helpful. Thank you and
best regards, Markus
Hi Andreas:
can you try the attached patch? Just add the line as shown in the patch. It should allow "PRACK" messages and make the stack reply to them correctly.
Thank you for this - the signalling should now work correctly, as the call connects immediately once the internal extension 221 (against which osmo-cc-sip-endpoint registers) is called. There is one issue remaining
- once connected, no audio is transmitted. I don't believe this is
related to this patch, though, as my Sipgate account continues working normally with the patch applied. As I am testing this in an openSUSE VM, I suspect it will be more of a firewall issue - or perhaps yet another Auerswald issue. I will be looking into this in more detail a little later today. I'm also attaching a SNGREP capture just in case this were helpful. Thank you and
I've also created an Auerswald trace. The relevant IP addresses are 10.1.2.1 for the PBX and 10.1.1.14 for the VM. I'm noticing a few ICMP failures (destination host unreachable) which could be indicative of a firewall problem in the VM, but I'm not really sure this actually causes no audio to be transmitted. I'm glad for any indication as to what this could be caused by, and I'll continue investigating the firewall issue also.
Best regards, Markus
Hi Andreas:
can you try the attached patch? Just add the line as shown in the patch. It should allow "PRACK" messages and make the stack reply to them correctly.
Thank you for this - the signalling should now work correctly, as the call connects immediately once the internal extension 221 (against which osmo-cc-sip-endpoint registers) is called. There is one issue remaining
- once connected, no audio is transmitted. I don't believe this is
related to this patch, though, as my Sipgate account continues working normally with the patch applied. As I am testing this in an openSUSE VM, I suspect it will be more of a firewall issue - or perhaps yet another Auerswald issue. I will be looking into this in more detail a little later today. I'm also attaching a SNGREP capture just in case this were helpful. Thank you and
It may not be a firewall issue after all - if I deactivate the openSUSE VM firewall completely, still no audio is transmitted. So I will try to explore some other possibilities, but am afraid I don't have any immediate leads now.
Thank you for your work and best regards, Markus
osmocom-analog@lists.osmocom.org
-
Andreas Eversberg -
atmega＠wp.pl -
Markus Stamm