I am still unable to connect with a audio call using pjsua & sip2sip.info service. I tried all udp/tcp options
my config file options are :
--use-ice
--use-turn
--stun-srv=stun.stunprotocol.org:3478
--turn-srv=numb.viagenie.ca:3479
--turn-user=xxxxx at yyyyyyy.com
--turn-passwd=xxxxxx
--registrar=sip:sip2sip.info
--outbound=sip:proxy.sipthor.net
--realm=sip2sip.info
--publish
--id=sip:uuuuu at sip2sip.info
--username=uuuuuuu
--password=zzzzzzz
--local-port=5091
I can make a test calls for sip:3333 at sip2sip.info, 4444 at sip2sip.info & sip:room at conference.sip2sip.info successfully. But when I make a outgoing call to another sip address, the call gets confirmed and I here voice saying ?the person you are calling is unavailable?. I can still make IM between each other.
I checked the firewall settings, those are OK. Thing is when I make a call using one pjsua console, the other pjsua console does not get incoming call notification. Cross checking the sip2sip.info server log tell that it goes to a ?BYE? message after connecting.
Can any on has an idea whats wrong. Are there any other settings to be done in the config-file for pjsua.
Regards,
Mohsin Z Barbhaiwala,
Manager - Design,
Spectrum Solutions & Technologies Pvt. Ltd.
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