I use this config from command line and it work with sip2sip, let try from your side: --id sip:xxx at sip2sip.info --registrar sip:sip2sip.info --realm sip2sip.info --username xxx --password xxx --outbound sip:proxy.sipthor.net --nameserver proxy.sipthor.net --log-level 3 --no-tcp From: pjsip [mailto:pjsip-bounces@xxxxxxxxxxxxxxx] On Behalf Of SSTPL (Design) Sent: Thursday, March 13, 2014 6:08 PM To: pjsip at lists.pjsip.org Subject: Still unable to connect with a audio call with pjsua & sip2sip.info Importance: High 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140313/0b5660d8/attachment-0001.html>
