When I dial out from my Asterisk 1.4.19 installation on Debian (three SIP hardphones on a LAN, and an IAX2 connection over DSL to a commercial trunk) I get this error on the console:
-- Executing [40618405@default:1] Set("SIP/21-081ceea8", "CALLERID(all)=nnnnnnnnnnnn <88821268>") in new stack
-- Executing [40618405@default:2] Dial("SIP/21-081ceea8", "IAX2/88821268/40618405|30|r") in new stack
[Sep 11 12:05:58] WARNING: app_dial.c:1202 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)
-- Executing [40618405@default:3] Congestion("SIP/21-081ceea8", "") in new stack
== Spawn extension (default, 40618405, 3) exited non-zero on 'SIP/21-081ceea8'
I can’t see any traffic on the wire using ngrep, and the registry looks good:
filserver*CLI> iax2 show registry
Host dnsmgr Username Perceived Refresh State
85.nnn.nnn.83:4569 N 88821268 85.nnn.nn.197:10000 60 Registered
85.nnn.nnn.82:4569 N 88821268 85.nnn.nn.197:10002 60 Registered
I can see traffic with ngrep while registering, and every 60 seconds after that.
That “no route to destination” error is causing my hair to thin, and my trunk provider tells me that it’s “usually something else”, and that the errormessage is not that descriptive.
What can I do to get more/better debugging info? I can’t figure out what’s wrong.
( my iax.conf and extensions.conf on http://pastebin.com/mb0020bd )