Hi,
I'm sorry for cross-posting this (from
http://forums.digium.com/viewtopic.php?t=64280), but I havn't got any
replies in the forum..
........
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 at default:1] Set("SIP/21-081ceea8",
"CALLERID(all)=nnnnnnnnnnnn <88821268>") in new stack
-- Executing [40618405 at default:2] Dial("SIP/21-081ceea8",
"IAX2/88821268/40618405|30|r") in new stack
[Sep 11 12:05:58] WARNING[7098]: 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 at 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.
Thanks!
- Martin
( my iax.conf and extensions.conf on http://pastebin.com/mb0020bd )
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Martin Seebach schrieb:> 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 at default:2] Dial("SIP/21-081ceea8", "IAX2/88821268/40618405|30|r") in new stack > [Sep 11 12:05:58] WARNING[7098]: 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 at default:3] Congestion("SIP/21-081ceea8", "") in new stack> 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.Maybe something is broken in recent versions of chan_iax2.c? http://lists.digium.com/pipermail/asterisk-users/2008-September/218560.html Not the same issue though. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 --
Hi, thanks for the reply , ----- Original Message ----- From: "Philipp Kempgen" <philipp.kempgen at amooma.de>> Maybe something is broken in recent versions of chan_iax2.c? > http://lists.digium.com/pipermail/asterisk-users/2008-September/218560.html > Not the same issue though.I doubt it. It has been working fine for a while, and others report IAX2 working fine. - Martin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080924/5f55b2d5/attachment.htm
On Wed, Sep 24, 2008 at 4:45 AM, Martin Seebach <mail at martinseebach.dk> wrote:> Hi, thanks for the reply, > > ----- Original Message ----- > From: "Philipp Kempgen" <philipp.kempgen at amooma.de> > >> Maybe something is broken in recent versions of chan_iax2.c? >> >> http://lists.digium.com/pipermail/asterisk-users/2008-September/218560.html >> Not the same issue though. > > I doubt it. It has been working fine for a while, and others report IAX2 > working fine. > > - Martin >Do a show codecs. Maybe IAX2 is not loaded. Did you build and load ztdummy (assuming you have no Zaptel/Dahdi cards? http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Thanks, Steve Totaro
Hi, ----- "Steve Totaro" wrote:> Do a show codecs.It looks right. ulaw is loaded, and that's the only thing I allow, on both SIP and IAX.> Maybe IAX2 is not loaded.Looks like it: filserver*CLI> module show like iax2 Module Description Use Count chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 1 modules loaded> Did you build and load ztdummy (assuming you have no Zaptel/Dahdi cards?No - but i don't use MeetMe? Thanks, Martin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080924/d7092d81/attachment.htm
> > > -- Executing [40618405 at default:1] Set("SIP/21-081ceea8", > "CALLERID(all)=nnnnnnnnnnnn <88821268>") in new stack > -- Executing [40618405 at default:2] Dial("SIP/21-081ceea8", > "IAX2/88821268/40618405|30|r") in new stack > [Sep 11 12:05:58] WARNING[7098]: 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 at 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.After looking at your iax.conf and extensions.conf I believe you are under the misconception that if you 'register' to a provider, then you can send and receive calls. The fact is that you 'register' to receive calls, but you must define a trunk in order to Dial Out. Your iax.conf [88821268] entry is not a trunk as you have not defined a host. That is why you get "cause 3 - No route to destination". Asterisk does not have any host defined in order to route that call. You need to talk to your provider for instructions on how to setup the trunk. Andres http://www.neuroredes.com> > Thanks! > > - Martin > > ( my iax.conf and extensions.conf on http://pastebin.com/mb0020bd ) > >------------------------------------------------------------------------ > >_______________________________________________ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >AstriCon 2008 - September 22 - 25 Phoenix, Arizona >Register Now: http://www.astricon.net > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >