I am really confused and hope someone can help. I have used this guide wiki.asterisk.org/wiki/display/ … P+Accounts to create two SIP Accounts and register my android phones (with SipDemo) to Asterisk. I use my laptop as WiFi-Hotspot and connect my phones. It works. But as soon as I try to make a call, I get the error message:
secret=101 ; put a strong, unique password here instead
secret=102 ; put a strong, unique password here instead[/quote]
Could you please help me?
You need to dial 6002, not user2.
Thanks a lot, david55.
May I ask you another question without opening a new topic? Now I can make audio calls using SIP, but when the call is answered there is only one way audio. I know (after reading some info), that it is probably the NAT problem. I tried to define externip, localnet & nat=yes in my sip.conf, but I still get one way audio. Do you have any idea, what else I can do to fix it?
Thanks in advance.
Please start a new thread, so that people can see there is a new subject. Preferably do so on Asterisk Support, which is where these questions should be.
You haven’t provided any protocol traces or indicated which direction isn’t working.
I solved this problem by defining the ports for RTP in rtp.conf like so:
And also port forwarded the above on my firewall to my asterisk box.