Velocity Reviews - Computer Hardware Reviews

Velocity Reviews > Newsgroups > Computing > UK VOIP > Asterisk AMP & Incoming calls

Reply
Thread Tools

Asterisk AMP & Incoming calls

 
 
Peter Watson
Guest
Posts: n/a
 
      05-25-2005
I've setup Asterisk on a Linux box that is running pptp (via eth0) to my
Speedtouch router. As a result the * box is 'visible' on a proper WAN
IP. My IP phones are on a 10.0.0.x network, connected to eth0.

I've setup two trunks - Call1899 (via iax) and Sipgate (via sip) and can
make outgoing calls via either route.

I'm using AMP to set all this up, but I can't get incoming calls (via
Sipgate) to work. It probably doesn't help that the AMP 'Incoming
calls' screen has greyed out radio buttons for selecting where incoming
calls should be routed to and that the 'active' selection is 'Ring
Group' but I've put my IP phone into Ring Group #1 and selected it. If
I dial 7777 (AMP configured demo extension that behaves as an incoming
call) the IP phone rings correctly but if I ring my Sipgate PSTN number
the call doesn't go through.

Has anyone else got this working or do I need to go back to editing the
..conf files by hand

I've posted a debug file at http://www.pwatson.org/asterisk_log.txt if
anyone has got time to wade through it and shed some light!

TIA

Peter
 
Reply With Quote
 
 
 
 
Ian
Guest
Posts: n/a
 
      05-25-2005

"Peter Watson" <(E-Mail Removed)> wrote in message
news:4294a89d$0$39077$(E-Mail Removed)...
> I've setup Asterisk on a Linux box that is running pptp (via eth0) to my
> Speedtouch router. As a result the * box is 'visible' on a proper WAN
> IP. My IP phones are on a 10.0.0.x network, connected to eth0.
>
> I've setup two trunks - Call1899 (via iax) and Sipgate (via sip) and can
> make outgoing calls via either route.
>
> I'm using AMP to set all this up, but I can't get incoming calls (via
> Sipgate) to work. It probably doesn't help that the AMP 'Incoming
> calls' screen has greyed out radio buttons for selecting where incoming
> calls should be routed to and that the 'active' selection is 'Ring
> Group' but I've put my IP phone into Ring Group #1 and selected it. If
> I dial 7777 (AMP configured demo extension that behaves as an incoming
> call) the IP phone rings correctly but if I ring my Sipgate PSTN number
> the call doesn't go through.
>
> Has anyone else got this working or do I need to go back to editing the
> .conf files by hand
>
> I've posted a debug file at http://www.pwatson.org/asterisk_log.txt if
> anyone has got time to wade through it and shed some light!
>
> TIA
>
> Peter

Hi
relevent bits of sip.conf and extensions.conf would be a bit more usefull
though.
personally I only edit the conf files. Seeing no need to use a gui

Ian


 
Reply With Quote
 
 
 
 
Ian
Guest
Posts: n/a
 
      05-25-2005

"Peter Watson" <(E-Mail Removed)> wrote in message
news:4294a89d$0$39077$(E-Mail Removed)...
> I've setup Asterisk on a Linux box that is running pptp (via eth0) to my
> Speedtouch router. As a result the * box is 'visible' on a proper WAN
> IP. My IP phones are on a 10.0.0.x network, connected to eth0.
>

Why didnt you just forward the relevent ports?
Its lot easier and tried and tested.


<snip>

IAn


 
Reply With Quote
 
Peter Watson
Guest
Posts: n/a
 
      05-25-2005

Ian wrote:

>
> Why didnt you just forward the relevent ports?
> Its lot easier and tried and tested.
>

Ah, that's a long story - I've just moved to Bulldog and my IP address
is now dynamic (it changes every time my ADSL connects). I also have a
dyndns domain so I'm running an update client to keep my IP address
updated in their DNS. Running pptp seemed like the easiest way of
making sure the update client gets the correct address I may go back
to a conventional setup and put the Linux box in the DMZ instead....

Peter
 
Reply With Quote
 
The Cable Guy
Guest
Posts: n/a
 
      05-25-2005
Peter Watson wrote:
|| I've setup Asterisk on a Linux box that is running pptp (via eth0)
|| to my Speedtouch router. As a result the * box is 'visible' on a
|| proper WAN IP. My IP phones are on a 10.0.0.x network, connected to
|| eth0.
||
|| I've setup two trunks - Call1899 (via iax) and Sipgate (via sip) and
|| can make outgoing calls via either route.
||
|| I'm using AMP to set all this up, but I can't get incoming calls (via
|| Sipgate) to work. It probably doesn't help that the AMP 'Incoming
|| calls' screen has greyed out radio buttons for selecting where
|| incoming calls should be routed to and that the 'active' selection
|| is 'Ring Group' but I've put my IP phone into Ring Group #1 and
|| selected it. If I dial 7777 (AMP configured demo extension that
|| behaves as an incoming call) the IP phone rings correctly but if I
|| ring my Sipgate PSTN number the call doesn't go through.
||
|| Has anyone else got this working or do I need to go back to editing
|| the .conf files by hand
||
|| I've posted a debug file at http://www.pwatson.org/asterisk_log.txt
|| if anyone has got time to wade through it and shed some light!
||
|| TIA
||
|| Peter

Yep.

Working here.

You've got to set the Sipgate number as a DID and point the DID to an
extension or ring group.

make sure your register for the Sipgate trunk is like this -
register=7DigitSIPNumber:8CHARACTERPASSWORD@sipgat e.co.uk/7DigitSipNumber.

and that your user details *&* peer details are like this-

authuser=7DigitSipNumber
context=ext-did
dtmfmode=info
fromdomain=sipgate.co.uk
fromuser=7DigitSipNumber
host=sipgate.co.uk
insecure=very
qualify=yes
secret=8CHARACTERPASSWORD
type=peer
username=7DigitSipNumber

context=ext-did

When you create your DID, make sure it is your 7DigitSipNumber & not your
geographic number that you use.

There are other ways & means to do what you want, I can imagine, however,
the above works fine here.

****************

The thing I need help on is installing Perl so that I can use Webmin to set
up Sendmail (actually I just want to set up Sendmail)


 
Reply With Quote
 
The Cable Guy
Guest
Posts: n/a
 
      05-25-2005
The Cable Guy wrote:
|| Peter Watson wrote:
|||| I've setup Asterisk on a Linux box that is running pptp (via eth0)
|||| to my Speedtouch router. As a result the * box is 'visible' on a
|||| proper WAN IP. My IP phones are on a 10.0.0.x network, connected
|||| to eth0.
||||
|||| I've setup two trunks - Call1899 (via iax) and Sipgate (via sip)
|||| and can make outgoing calls via either route.
||||
|||| I'm using AMP to set all this up, but I can't get incoming calls
|||| (via Sipgate) to work. It probably doesn't help that the AMP
|||| 'Incoming calls' screen has greyed out radio buttons for selecting
|||| where
|||| incoming calls should be routed to and that the 'active' selection
|||| is 'Ring Group' but I've put my IP phone into Ring Group #1 and
|||| selected it. If I dial 7777 (AMP configured demo extension that
|||| behaves as an incoming call) the IP phone rings correctly but if I
|||| ring my Sipgate PSTN number the call doesn't go through.
||||
|||| Has anyone else got this working or do I need to go back to editing
|||| the .conf files by hand
||||
|||| I've posted a debug file at http://www.pwatson.org/asterisk_log.txt
|||| if anyone has got time to wade through it and shed some light!
||||
|||| TIA
||||
|||| Peter
||
|| Yep.
||
|| Working here.
||
|| You've got to set the Sipgate number as a DID and point the DID to an
|| extension or ring group.
||
|| make sure your register for the Sipgate trunk is like this -
||
register=7DigitSIPNumber:8CHARACTERPASSWORD@sipgat e.co.uk/7DigitSipNumber.
||
|| and that your user details *&* peer details are like this-
||
|| authuser=7DigitSipNumber
|| context=ext-did
|| dtmfmode=info
|| fromdomain=sipgate.co.uk
|| fromuser=7DigitSipNumber
|| host=sipgate.co.uk
|| insecure=very
|| qualify=yes
|| secret=8CHARACTERPASSWORD
|| type=peer
|| username=7DigitSipNumber
||
|| context=ext-did
||
|| When you create your DID, make sure it is your 7DigitSipNumber & not
|| your geographic number that you use.
||
|| There are other ways & means to do what you want, I can imagine,
|| however, the above works fine here.
||
|| ****************
||
|| The thing I need help on is installing Perl so that I can use Webmin
|| to set up Sendmail (actually I just want to set up Sendmail)

Bad form to reply to my own post, I know.

I assumed you would clearly understand to replace 7DigitSipNumber with your
actual 7 digit sip number etc.....


 
Reply With Quote
 
Roly
Guest
Posts: n/a
 
      05-25-2005
"Peter Watson" <(E-Mail Removed)> wrote in message
news:4294a89d$0$39077$(E-Mail Removed)...
> I've setup Asterisk on a Linux box that is running pptp (via eth0) to my
> Speedtouch router. As a result the * box is 'visible' on a proper WAN IP.
> My IP phones are on a 10.0.0.x network, connected to eth0.
>
> I've setup two trunks - Call1899 (via iax) and Sipgate (via sip) and can
> make outgoing calls via either route.
>
> I'm using AMP to set all this up, but I can't get incoming calls (via
> Sipgate) to work. It probably doesn't help that the AMP 'Incoming calls'
> screen has greyed out radio buttons for selecting where incoming calls
> should be routed to and that the 'active' selection is 'Ring Group' but
> I've put my IP phone into Ring Group #1 and selected it. If I dial 7777
> (AMP configured demo extension that behaves as an incoming call) the IP
> phone rings correctly but if I ring my Sipgate PSTN number the call
> doesn't go through.
>
> Has anyone else got this working or do I need to go back to editing the
> .conf files by hand
>
> I've posted a debug file at http://www.pwatson.org/asterisk_log.txt if
> anyone has got time to wade through it and shed some light!
>
> TIA
>
> Peter


Here are the two screenshots from my setup. The bit that miffed me for a
while was that seemingly a DID route needs to be defined.

http://makeashorterlink.com/?D1C71242B
and
http://makeashorterlink.com/?L2E71542B

Roly.


 
Reply With Quote
 
Peter Watson
Guest
Posts: n/a
 
      05-25-2005
Ian wrote:

>
> Hi
> relevent bits of sip.conf and extensions.conf would be a bit more usefull
> though.
> personally I only edit the conf files. Seeing no need to use a gui
>

I've now uploaded the .conf files as http://www.pwatson.org/conf_files.zip

Thanks,

Peter
 
Reply With Quote
 
Ian
Guest
Posts: n/a
 
      05-25-2005

"Peter Watson" <(E-Mail Removed)> wrote in message
news:4294b834$0$557$(E-Mail Removed)...
> Ian wrote:
>
> >
> > Hi
> > relevent bits of sip.conf and extensions.conf would be a bit more

usefull
> > though.
> > personally I only edit the conf files. Seeing no need to use a gui
> >

> I've now uploaded the .conf files as http://www.pwatson.org/conf_files.zip
>
> Thanks,
>

Only had time to skim them, But cant see any mention of incoming sipgate
number in the extensions.conf. it need to be in there it wont be handled by
the s extension

Ian


 
Reply With Quote
 
Peter Watson
Guest
Posts: n/a
 
      05-26-2005
In article <4294b7eb$0$580$(E-Mail Removed)>,
http://www.velocityreviews.com/forums/(E-Mail Removed) says...

>
> Here are the two screenshots from my setup. The bit that miffed me for a
> while was that seemingly a DID route needs to be defined.
>
> http://makeashorterlink.com/?D1C71242B
> and
> http://makeashorterlink.com/?L2E71542B
>
> Roly.
>
>

Thanks for all the relplies to this thread - I've altered my outgoing
registration stuff and now incoming calls work!

Peter
 
Reply With Quote
 
 
 
Reply

Thread Tools

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Trackbacks are On
Pingbacks are On
Refbacks are Off


Similar Threads
Thread Thread Starter Forum Replies Last Post
Messagenet incoming number setup on Asterisk@home paul123 UK VOIP 4 03-08-2010 02:07 PM
#######modernpractice.webs.com&&&&&&& sharmi3435@gmail.com Python 0 04-06-2009 06:24 AM
Voipdiscount Outgoing Calls with Sipgate Incoming Calls on a Linksys PAP2 Giganews UK VOIP 27 10-09-2006 11:28 PM
The connection to the server has failed. Account: 'incoming.yahoo.verzon.net', Server: 'incoming.yahoo.verizon.net', Protocol: POP3, Port: 110, Secure(SSL): No, Socket Error: 10061, Error Number: 0x800CCC0E Michael Bower Computer Support 3 10-01-2006 03:44 PM
Asterisk@Home - Incoming Calls Sean UK VOIP 10 10-03-2005 06:09 PM



Advertisments