Hi all, I’m new to Asterisk and can’t get my head around this error. I’m using diax & trying to call one PC from another across the LAN
-- Accepting AUTHENTICATED call from 10.10.10.230:
> requested format = gsm,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (),
> priority = mine
-- Executing Dial("IAX2/james@james-3", "IAX2/1001") in new stack
Aug 4 23:25:45 WARNING[11777]: chan_iax2.c:2661 create_addr: No such host: 1001
Aug 4 23:25:45 NOTICE[11777]: app_dial.c:1092 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
Aug 4 23:25:56 WARNING[11777]: pbx.c:2337 __ast_pbx_run: Timeout, but no rule 't' in context 'default'
-- Hungup 'IAX2/james@james-3'
Asterisk seems to know about the registered users
*CLI> iax2 show peers
Name/Username Host Mask Port Status
andrew 10.10.10.2 (D) 255.255.255.255 4569 Unmonitored
james 10.10.10.230 (D) 255.255.255.255 4569 Unmonitored
2 iax2 peers [0 online, 0 offline, 2 unmonitored]
I am running a very simple config and have stripped down iax & extensions.conf in an attempt to solve the problem
…extensions.conf…
[code][general]
static=yes
writeprotect=no
[globals]
[default]
exten => 1001,1,Dial(IAX2/1001)
exten => 1002,1,Dial(IAX2/1002)
[/code]
…iax.conf…
[code][general]
bindport=4569
allow=all
[james]
type=friend
host=dynamic
auth=md5
secret=pass
host=dynamic
context=default
[andrew]
type=friend
host=dynamic
auth=md5
secret=pass
host=dynamic
context=default[/code]
I have also tried a SIP setup with sjphone and have had the same results - any help would be appreciated
raz
- helps if I’d read the