Problem Register with CallManager

[Claudiocre]

Hi all, I have a problem with sipcmd and CallManager

It appears that during the negotiation phase, the client does not keeps the registration and therefore during call. From sniffer, I don't see the registration on Server: "Not Found "

09:52:53.011217 IP localhost.localdomain.sip > adriacm.agsdom.loc.sip: SIP: REGISTER sip:192.168.71.253 SIP/2.0 09:52:53.016301 IP adriacm.agsdom.loc.sip > localhost.localdomain.sip: SIP: SIP/2.0 100 Trying 09:52:53.019156 IP localhost.localdomain.sip > adriacm.agsdom.loc.sip: SIP: INVITE sip:0039339XXXXXX@192.168.71.253 SIP/2.0 09:52:53.019173 IP localhost.localdomain > adriacm.agsdom.loc: udp 09:52:53.025153 IP adriacm.agsdom.loc.sip > localhost.localdomain.sip: SIP: SIP/2.0 100 Trying 09:52:53.025312 IP adriacm.agsdom.loc.sip > localhost.localdomain.sip: SIP: SIP/2.0 503 Service Unavailable 09:52:53.026272 IP localhost.localdomain.sip > adriacm.agsdom.loc.sip: SIP: ACK sip:0039339XXXXXX@192.168.71.253 SIP/2.0 09:52:53.122499 IP adriacm.agsdom.loc.sip > localhost.localdomain.sip: SIP: SIP/2.0 401 Unauthorized 09:52:53.124501 IP localhost.localdomain.sip > adriacm.agsdom.loc.sip: SIP: REGISTER sip:192.168.71.253 SIP/2.0 09:52:53.128381 IP adriacm.agsdom.loc.sip > localhost.localdomain.sip: SIP: SIP/2.0 100 Trying 09:52:53.131785 IP adriacm.agsdom.loc.sip > localhost.localdomain.sip: SIP: SIP/2.0 404 Not Found

I have tried the userSIP with other clients and it works.

There are the commands, the log and output from call.

COMMAND sipcmd -P sip -u "UserSIP" -c "Password2022." -a "329" -w "192.168.71.253" -o "/root/sip.log" -x "c00339XXXXXX;ws500;vAllarmeRosso.wav;ws15000;h"

I have installed module on Centos7 (with 1 interface (eth0)) and there are no nat

LOG sip.log

OUTPUT `Starting sipcmd in debug mode Manager Init 0:00.025 Opal Garba...cf736fe700 PTLib Started thread 0x1458770 (7831) Opal Garbage:0x7fcf736fe700 initialising SIP endpoint... TestChanAudio TestChanAudio Listening for SIP signalling on 0.0.0.0:5060 SIP listener up registered as sip:UserSIP@192.168.71.253 Created LocalEndPoint Main

Call

TestPhone::Main: calling "0039339XXXXXX" using gateway "192.168.71.253" at Fri Oct 14 10:11:09 2022

Setting up a call to: sip:0039339XXXXXX@192.168.71.253 LocalEndpoint::MakeConnection LocalEndpointCreateConnection LocalConnection OnIncomingConnection: token=Lb12481712 available codes G.722-64k G.722.1-24k G.722.1-32k G.722.2 SpeexIETFWide-20.6k SpeexWB SpeexWide-20.6k G.711-ALaw-64k G.711-uLaw-64k G.726-16k G.726-24k G.726-32k G.726-40k GSM-06.10 GSM-AMR LPC-10 MS-GSM MS-IMA-ADPCM SpeexIETFNarrow-11k SpeexIETFNarrow-15k SpeexIETFNarrow-18.2k SpeexIETFNarrow-24.6k SpeexIETFNarrow-5.95k SpeexIETFNarrow-8k SpeexNB SpeexWNarrow-8k iLBC iLBC-13k3 iLBC-15k2 UserInput/RFC2833 NamedSignalEvent H.261 RFC4175_YCbCr-4:2:0 theora MSRP SIP-IM T.140 H.224/H323AnnexQ used codes G.722-64k G.722.1-24k G.722.1-32k G.722.2 SpeexIETFWide-20.6k SpeexWB SpeexWide-20.6k G.711-ALaw-64k G.711-uLaw-64k G.726-16k G.726-24k G.726-32k G.726-40k GSM-06.10 GSM-AMR LPC-10 MS-GSM MS-IMA-ADPCM SpeexIETFNarrow-11k SpeexIETFNarrow-15k SpeexIETFNarrow-18.2k SpeexIETFNarrow-24.6k SpeexIETFNarrow-5.95k SpeexIETFNarrow-8k SpeexNB SpeexWNarrow-8k iLBC iLBC-13k3 iLBC-15k2 UserInput/RFC2833 NamedSignalEvent H.261 RFC4175_YCbCr-4:2:0 theora MSRP SIP-IM T.140 H.224/H323AnnexQ connection set up to sip:0039339XXXXXX@192.168.71.253 TestPhone::Main: calling "sip:0039339XXXXXX@192.168.71.253" for 0 OnReleased: reason: Unknown OnReleased: reason: Unknown OnClearedCall

Wait: waiting for 500ms

Wait: wait done

Voice audiofile=AllarmeRosso.wav

PlaybackAudioFile PlaybackAudioFile: state 3

Wait: waiting for 15000ms

~LocalConnection Wait: wait done

Hangup

Hangup: at Fri Oct 14 10:11:25 2022

TestPhone::Main: shutting down TestPhone::Main: exiting... Exiting... ~Manager`

[Claudiocre]

I also highlight:

I have recevid : OnReleased: reason: Unknown and a SIP/2.0 503 Service Unavailable

[soroshsabz]

@Claudiocre are you sure, your SIP Server work correctly? did you can, test your server with other tools, to make sure all functionality works correctly?

thanks

[Claudiocre]

Hi, the server is work correctly, it's manages all phone of the my society. And now I can not onother test.