I'm testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also plays tt-weasels as it's supposed to, but it's not the tt-weasels under its extension. It also dials the s extension. I only placed the s extension in the dial plan to verify that the traffic was going across the IAX trunk and hitting the correct context. Any help…
Trying to upgrade some call servers, in the lab making sure all my
applications work, ran into an issue with some manager perl scripts
that pull and reset database info, it seems the command and result
responses have changed but I'm not sure how to resolve. My scripts
are using CPAN Asterisk::Manager; and are working fine on asterisk
1.2.32 but not on Asterisk 18.104.22.168. Here is the abbreviated script where 1.2.32 is astman1 and 22.214.171.124 is astman2:
I've just upgrade from 126.96.36.199 to 188.8.131.52 and some kind of endpoint
aren't able to register. Message is:
[Jul 16 01:26:15] NOTICE: chan_sip.c:23511
handle_request_register: Registration from '
'X.X.X.X:5060' - No matching peer found sip.conf [user637801]
call-limit=2 REGISTER sip:Y.Y.Y.Y SIP/2.0.
CSeq: 2 REGISTER.
Via: SIP/2.0/UDP 192.168.100.2:42904;branch=z9hG4bK-508b5-13aa048e-5ff4a5b9.
I am trying to get 2 accounts from voipblaster to talk to each other.
Calls withing voipblaster network is free. If I configure two sip clients with
the two accounts it works fine
however with Asterisk I am getting SIP 401
In my Sip.conf file I under general
register = user:email@example.com
then I have a sip peer
username = kiks2010
secret = password
fromuser = kiks2010