Hi,
I change sip to pjsip. Before when i used sip i could set verbosity to level 3 and i able to saw actual program. Now, after that when i changed configuration from sip to pjsip and i able to register my endpoints i would like to make call to asterisk and hear any sounds. But thereis nothing. Any sugest what i done wrong ?
The channel technology driver has no effect on the logging in pbx.c.
Ok, so you told me that if I will use pjsip then i could’t see any dialplan progress? And this is possible only if i using chan_sip? Am I right ?
I have another question. There is very simple dialplan:
[context]
exten => _X.,1,Noop(testing)
same => n,Answer
same => n,Playback(tt-weseals)
same => n,Hangup
So, my calls should be answered and i could hear any audio, but it is not happened. In sip signalling i saw Invite, authentication and trying is last thing. There is no either sip ok or sip bye
No David doesn’t mean this. Logging has nothing to do with channel technology, be it SIP,PJSIP, Dahdi or something else.
Did you check your configuration (logger.conf,asterisk.conf) for logging?
–Satish Barot
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