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SIPGATE Inbound Asterisk lady announcing “The number you have dialled is not in service” Problem

wayne.canavan@n...
Posts: 5
 Member Since:
2007-05-20
Submitted by wayne.canavan@n... on Sun, 05/27/2007 - 12:50pm.

Hi Everyone

I have spent a good couple of days trying to figure this out and rebuilt the TrixBox twice to make sure I haven’t mucked up the conf files by editing them directly.

On a nice, new fresh TrixBox 2.2 I still have the same problem, can anyone help?

Background History:
I have setup a TrixBox 2.2 running under VMWARE for evaluation purposes. I have followed the tutorial “Getting Started with TrixBox tutorial.” from the website http://asterisktutorials.com

So far so good, I have my IVR setup and answering calls when I perform a test using the extension “7777”; even the press “1” for Blah Blah Blah and “2” for XYZ works.

I have purchased some books on the subject PACKT (Barrie Dempster & Kerry Garrison) although these are more in-depth and seem to be more relevant to the inner workings of Asterisk.

I know my SIPGATE is working because I can hear the Asterisk lady announcing “The number you have dialled is not in service” so this has to be a routing issue.

Where do I go from hear?

Regards
Wayne

‹ Ive been jacked by fonality When one peer of an active channel unexpected loses the connection, the channel keeps alive ›

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josegaal
Posts: 34
 Member Since:
2007-03-22
Hi, try this link, I had a
Sun, 05/27/2007 - 1:15pm

Hi, try this link, I had a similar problem and this article helped a lot:

http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_...


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wayne.canavan@n...
Posts: 5
 Member Since:
2007-05-20
I may have narrowed down the problem to SIP authentication.
Sun, 05/27/2007 - 1:45pm

Hi Everyone

Thanks for the response, although I may have narrowed down the problem to SIP authentication.

Under the General Settings, I scrolled down to the section called “Security” and set the box Allow Anonymous Inbound SIP Calls? to “YES” and it worked.

So, do I have another problem to do with authentication?

Regards
Wayne


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linker3000
Posts: 110
 Member Since:
2006-06-08
Check your sip.conf looks
Sun, 05/27/2007 - 2:55pm

Check your sip.conf looks something like this - assuming you have a fixed WAN IP address:

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.

[general]
bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
;Put your supported CODEC list below...
allow=g729
allow=ulaw
allow=alaw
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
;context = from-sip-external ; Send unknown SIP callers to this context
context = from-trunk
callerid = Unknown
tos=0x68

;next is your WAN's fixed IP address:
externip = 12.34.56.78
;next is your local network address:
localnet = 192.168.101.0/255.255.255.0
;What to dial for voicemail:
vmexten = *97

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf


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wayne.canavan@n...
Posts: 5
 Member Since:
2007-05-20
IT Works - Yip eeeeeeeeeee
Mon, 05/28/2007 - 2:39am

Hi Linker3000

Thank you so much for the reply I really appreciated it, and following what you had suggested worked perfectly when I switched it back to Allow Anonymous Inbound SIP Calls? to “NO”

It has been a great learning experience so far and purchasing the PDF Ebooks has also helped me to understand a little more. I will keep my eye out for other Sipgate sufferers and try and repay your effort back in me helping others, as I get more knowledgeable.

It may take a while before you see me posting help replies, as they say a little knowledge can be dangerous!

Regards
Wayne


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harrymrh
Posts: 25
 Member Since:
2007-06-02
What does it do then
Fri, 06/29/2007 - 10:26am

Does changing the location of the inbound calls achieve something different to not allowing anonymous inbound calls. What makes a call anonymous or not? How might an incoming sip trunk be recognised and vailidated?
In my setup if I include the sip_nat.conf lines about my external and internal ip addresses I get an "all circuits are busy". Without it I get two way sipgate efficacy as long as anonymous calls are accepted. Any clues?
I am behind a draytek 2600 router with ports 5060 10001-20000 redirected to my trixbox local ip from the public side of the firewall.


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murfett-au
Posts: 1
 Member Since:
2009-01-05
Allowing Anonymous SIP fixed this for me also
Mon, 01/05/2009 - 5:22am

I had the same problem, and setting "Allow Anonymous Inbound SIP Calls?" to "Yes" in "Basics - General Settings - Security Settings" fixed this right up!


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