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Problem Register with CallManager #77

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Claudiocre opened this issue Oct 14, 2022 · 3 comments
Open

Problem Register with CallManager #77

Claudiocre opened this issue Oct 14, 2022 · 3 comments

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@Claudiocre
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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:[email protected] 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:[email protected] 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:[email protected]
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:[email protected]
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:[email protected]
TestPhone::Main: calling "sip:[email protected]" 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
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I also highlight:

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

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

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