asterisk users - Feb 2011 - Error: Unable to create channel of type 'SIP'

Hi,

I am using Trixbox 2.6.2.3, ISO install

I am getting the below error in `/var/log/asterisk/full`

Unable to create channel of type 'SIP' (cause 3 - No route to
destination)

Is there anyway to figure out which extension is causing this error ?

Thank you.

Best regards,
Sanjay

On Mon, Feb 7, 2011 at 7:42 AM, RSCL Mumbai <rscl.mumbai at gmail.com>
wrote:
> Hi,
>
> I am using Trixbox 2.6.2.3, ISO install
>
> I am getting the below error in `/var/log/asterisk/full`
>
> Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
>
> Is there anyway to figure out which extension is causing this error ?
>
> Thank you.
>
> Best regards,
> Sanjay
>
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Now,Sanjay, don't take this personally, you just happen to be ANOTHER person
who has sent an email to the list lately that just crosses my tolerance for
lack of respect for the "try and figure it out yourself before you ask for
help" mentality behind this (and most Open Source project's) mailing
list.....
'
First solution (1.5 seconds after I read your question)...check your call
detail records! ,You'll see the failed call(s)!

Second solution (thought of nanoseconds after the first one)....Now....I
want you to think just a tiny bit here...If you wanted to know if a host was
reachable, what would you do?.........This is the same thing, except you
have a list of hosts and you need to determine WHICH one cannot be
reached......You try to contact each host until you find one or more that
gives you a "no route to host" message!!!!!

ping is your friend, so is mtr, also a telnet session (over the port
specified for SIP to that host in your config) could be used......

Third possible method: What level of verbosity is the server currently
running at? If it's not running at 3 or higher, set verbose to at least 3.
That way you will see the dialplan executions that occur just before that
message. Once you see that, you'll most likely have your answer.

Useful tip: Another thing you could do, set qualify=yes on your sip
endpoints' configurations, since this is a "no route to host"
issue, you'll
see failure on at least one of them, which will also give you your
answer.....

Now, I'm going to sound like a jerk, but these are all simple methods that
you could/should have come up with...How many seconds did you spend thinking
about the issue before you decided to ask the list for help with a question
that is admittedly something you should have SOME idea regarding how to
test....

Man, I'm starting to just get pissed...That's what, 3 questions I've
seen in
the last 12 or less hours where the person asking the question OBVIOUSLY
doesn't want to put forth any effort on their own before asking the rest of
us how to do something?

Asterisk Documentation is your friend!
UNDERSTANDING at least 25% of how VoIP works is handy!
GOOGLE is your friend!

And in the name of any and all things/beings that you guys find to be holy,
put forth some damned effort before asking everyone else to do the work for
you!!!!

Finally, if you HAVE put forth effort, LET US KNOW!!! It lessens the chance
of you getting flamed by some guy who's been working for over 40 hours
STRAIGHT and is just tired of seeing email after email after email
containing questions that have been answered hundreds of times on the list
and there are readily available answers via documentation and/or a little
friggin googling......


That's it...I'm going back to barely reading the list...Every time I try
to
start reading it on a fairly often basis (in the hopes of being able to help
people with continuing issues AFTER putting some damn effort towards the
problem), I start seeing that 75-80% of new requests have 0-5% effort put
forth into trying to fix it themselves, and this includes basic stuff like
RTFM!!!!!

Cheers
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