Hey,
with chan_sip i could to this:
exten => xxxx,1,Dial(SIP/xxxx@sip.linphone.org,30)
And i had a linphone registered on the sip.linphone.com service, and it started ringing, no issues
Now, switched over to pjsip…
exten => xxxx,1,Dial(PJSIP/xxxx@sip.linphone.org,30)
and i get:
Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
Why is that? what am i missing?
Dial strings are specific to the channel driver. PJSIP is different[1]. This would be dialing a SIP URI case.
[1] Dialing PJSIP Channels - Asterisk Project - Asterisk Project Wiki
hmm, do i need to take this example then?
exten => _9NXXNXXXXXX,1,Dial(PJSIP/mytrunk/sip:${EXTEN:1}@203.0.113.1:5060)
But i dont have a trunk? on chan_sip it was not needed?
And chan_pjsip isn’t chan_sip. You must specify an endpoint, as it is what defines the configuration to use for the call (codecs for example).
hmm, still dont understand? this is my endpoint “xxxx@sip.linphone.org”
that endpoint i was able to call directly when i used chan_sip , still dont know why it cant be called when using pjsip ?
That’s a SIP URI. An endpoint in this case is a named entity from pjsip.conf of type endpoint. It defines the configuration to use for the call. You could have a generic “outgoing” one that defines your default configuration to use for outgoing calls to SIP URIs.
In chan_sip you could throw most things into the “general” section instead and have it use that configuration. Not the case with chan_pjsip.
its it much asked to provide me an example config? sorry a bit new to me
[outgoing]
type=endpoint
allow=!all,ulaw
Then to use:
PJSIP/outgoing/sip:${EXTEN}@sip.linphone.org
aha, lets try that
thnx, much appreciated!!
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