[Asterisk-Users] Sometimes yes - sometimes no (dialplan)

[Tzafrir Cohen]

On Mon, Jul 04, 2005 at 05:19:39PM +0800, Ronald_Wiplinger wrote:
> Robert Goodyear wrote:
> 
> >>>>>>I am confused about one of my installed server
> >>>>>>
> >>>>>>The dial plan seems to be ok, but sometimes NOTHING happens if I 
> >>>>>>try to dial an extension (from X-Lite), next time it is done.
> >>>>>>
> >>>>>>X-Lite does not have a tone, nothing and does also have no time 
> >>>>>>out. It seems it is not connected to the server. However, a sip 
> >>>>>>show users / sip show peers   shows that the phone is connected.
> >>>>>
> >>>>>SIP clients generate their own dialtone, so if you've got no tone, 
> >>>>>that sounds suspicious of a problem with the client itself. I 
> >>>>>assume you've debugged the problem by registering a hard SIP 
> >>>>>client on that server?
> >>>>
> >>>>The CLI prompt does not show anything either. It is like the phone 
> >>>>is not talking to asterisk at all.
> >>>>sip show users/peers   does show the phone.
> >>>
> >>>...shows the phone REGISTERED, yes?
> >>
> >>yes!!!
> >
> >
> >...yet no other information in the CLI or logs? C'mon, help us help 
> >you. The clue is in the question.
> 
> 
> I cannot make up a CLI entry ;-)
> There is nothing about it!!!
> As I said it is like it is not connected!!!!!

How do you know it is not connected?
Why do you assume it should be connected?

Please answer your questions, and while you do: verify all of your
assumptions. After you've answered them, please try to guess what our
next question would have been.

Is there a sip peer or it in sip.conf? How does that sip peer appear in
'sip show peers' on the CLI?

voip-info,org, google, and such are valueble resources for answering the
questions.

For example: "connected" basically means (for a SIP client) being
registered as a SIP peer. Though a client can technically connect
without registrating in advance.

So: is there a section for it in sip.conf? How does it appear in the
output of 'sip show peers'?

Do you have any reason to believe that the grandstream phone is actually
sending any packets to your asterisk computer?

Try running:

  tcpdump -n 'host IP_ADDRESS_THE_PHONE'

on your asterisk system. Ethereal may be useful for protocol analisys,
but tcpdump is great if you just want to know "if there is traffic". 

Naturally another thing to try is to eliminate one part of the problem:
can you use a different SIP client with the same definitions of the
server (or vice-versa)? Does that SIP client work with any other SIP
server?

-- 
Tzafrir Cohen         | tzafrir at jbr.cohens.org.il | VIM is
http://tzafrir.org.il |                           | a Mutt's  
tzafrir at cohens.org.il |                           |  best
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