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AsteriskWin32 - Incomming Calls Problem



 
 
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  #1  
Old May 5th 06, 12:50 PM posted to uk.telecom.voip
lab
external usenet poster
 
Posts: 10
Default AsteriskWin32 - Incomming Calls Problem

Hi

I wondered if anybody has seen this issue before and could set me on the
right way of correcting it. I have setup asterisk win32 on a colocated
machine, and setup some simple users/extensions.
If i place a call from a user/user that is logged in, it works.
If i place a call from a user/sip peer it works. (i have sipgate/voipfone
setup) dialing 8 first uses sipgate, dialing 9 first uses voipfone.
However when i try to get incomming calls working, they dont ever appear to
get through, on sipgate, i just get the busy tone/or number not recognized,
for voipfone i just get the voipfone answermachine, regardless if i dial
that by PSTN/SIP.

If i check my "sip show registry" i get
Host Username Refresh State
voipfone.co.uk:5060 301xxxxx 45 Registered
sipgate.co.uk:5060 98xxxxx 105 Registered

The machine is not behind a nat router and all firewall ports are open to
the relevant IP's (also there is no data being blocked on the firewall as i
have checked the logs), what seems to be strange is that after about 20-30
minutes of starting asterisk, both incomming numbers will work, but then an
hour or so later, they go off permantly again. Also if i check
sipgate/voipfone logs (from their websites) i see that they have made a note
of the missed calls, so it really does look like a problem with my setup, im
guessing a registration issue?

Can anybody help? Im guessing i should just bite the bullet and install
the latest linux versions, but surely the windows version should work
anyway? Thanx for any advice



  #2  
Old May 5th 06, 06:16 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 88
Default AsteriskWin32 - Incomming Calls Problem

lab wrote:
|| Hi
||
|| I wondered if anybody has seen this issue before and could set me on
|| the right way of correcting it. I have setup asterisk win32 on a
|| colocated machine, and setup some simple users/extensions.
|| If i place a call from a user/user that is logged in, it works.
|| If i place a call from a user/sip peer it works. (i have
|| sipgate/voipfone setup) dialing 8 first uses sipgate, dialing 9
|| first uses voipfone.
|| However when i try to get incomming calls working, they dont ever
|| appear to get through, on sipgate, i just get the busy tone/or
|| number not recognized, for voipfone i just get the voipfone
|| answermachine, regardless if i dial that by PSTN/SIP.
||
|| If i check my "sip show registry" i get
|| Host Username Refresh State
|| voipfone.co.uk:5060 301xxxxx 45 Registered
|| sipgate.co.uk:5060 98xxxxx 105 Registered
||
|| The machine is not behind a nat router and all firewall ports are
|| open to the relevant IP's (also there is no data being blocked on
|| the firewall as i have checked the logs), what seems to be strange
|| is that after about 20-30 minutes of starting asterisk, both
|| incomming numbers will work, but then an hour or so later, they go
|| off permantly again. Also if i check sipgate/voipfone logs (from
|| their websites) i see that they have made a note of the missed
|| calls, so it really does look like a problem with my setup, im
|| guessing a registration issue?
||
|| Can anybody help? Im guessing i should just bite the bullet and
|| install the latest linux versions, but surely the windows version
|| should work anyway? Thanx for any advice

Have you set up an inbound route for your numbers?

Asterisk normally sends unknown sip callers to "congestion"

You can either change the context sip calls come in on, or you could simply
create a DID for your number. The DID for Sipgate should be your 7 digit SIP
ID, not your PSTN number.

Also, what does your register string look like?

I would bite the bullet & install a proper Linux version, if I were you.


  #3  
Old May 5th 06, 06:56 PM posted to uk.telecom.voip
lab
external usenet poster
 
Posts: 10
Default AsteriskWin32 - Incomming Calls Problem


"Jono" wrote in message
k...

Have you set up an inbound route for your numbers?

Asterisk normally sends unknown sip callers to "congestion"

You can either change the context sip calls come in on, or you could
simply create a DID for your number. The DID for Sipgate should be your 7
digit SIP ID, not your PSTN number.

Also, what does your register string look like?


My registration strings are as follows

register = /3012xxxx
register = :5061/987xxxx

sometimes i have sipgate without any port , specifying 5061 appeared to help
a little. Obviously replacing the xxxx with the last 4 digits of my userid
and replacing sippass, with the proper sippassword (not my signup password
to the sites), if i place the same details into my PAP2, then everything
works fine.

My extensions appear to be setup correctly also, as sometimes (without
reloading/changing the config) some calls will get through and properly
routed to the correct SIP-USER etc.


I would bite the bullet & install a proper Linux version, if I were you.


I know and i will, its just i have only recently cleared out all my old
systems (so i have nothing at home, cept my main pc) and i would really like
it on my colocated server using vmware, which would need an upgrade (due in
2-3 months), so i cant do it on that at the moment. Even if i cant work out
what it is, it was worthwhile just from the learning experience. Thanx for
the reply


 




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