You signed in with another tab or window. Reload to refresh your session.You signed out in another tab or window. Reload to refresh your session.You switched accounts on another tab or window. Reload to refresh your session.Dismiss alert
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.
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
@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?
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`
The text was updated successfully, but these errors were encountered: