Extension not found in context.

[justuser]

Hello.

I am really confused and hope someone can help. I have used this guide https://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:

Code: Select all

[Aug  3 09:23:31] NOTICE[19841]: chan_sip.c:22753 handle_request_invite: Call from 'user1' (192.168.....) to extension 'user2' rejected because extension not found in context 'from-internal'

sip.conf:

[general]
transport=udp

[friends_internal](!)
type=friend
host=dynamic
context=from-internal
disallow=all
allow=ulaw
allowexternaldomains=yes

[user1](friends_internal)
secret=101 ; put a strong, unique password here instead

[user2](friends_internal)
secret=102 ; put a strong, unique password here instead

extensions.conf:

[from-internal]
exten=>6001,1,Dial(SIP/user1,20)
exten=>6002,1,Dial(SIP/user2,20)

Could you please help me?

[david55]

You need to dial 6002, not user2.

[justuser]

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.

[david55]

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.

[masterchan]

Hi justuser

I solved this problem by defining the ports for RTP in rtp.conf like so:

[general]
rtpstart=10000
rtpend=20000

And also port forwarded the above on my firewall to my asterisk box.