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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

Asterisk AMP & Incoming calls



 
 
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  #1  
Old May 25th 05, 05:32 PM posted to uk.telecom.voip
Peter Watson
external usenet poster
 
Posts: 70
Default Asterisk AMP & Incoming calls

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
  #2  
Old May 25th 05, 05:45 PM posted to uk.telecom.voip
Ian
external usenet poster
 
Posts: 289
Default Asterisk AMP & Incoming calls


"Peter Watson" wrote in message
...
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


  #3  
Old May 25th 05, 05:52 PM posted to uk.telecom.voip
Ian
external usenet poster
 
Posts: 289
Default Asterisk AMP & Incoming calls


"Peter Watson" wrote in message
...
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


  #4  
Old May 25th 05, 06:04 PM posted to uk.telecom.voip
Peter Watson
external usenet poster
 
Posts: 70
Default Asterisk AMP & Incoming calls


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
  #5  
Old May 25th 05, 06:27 PM posted to uk.telecom.voip
The Cable Guy
external usenet poster
 
Posts: 55
Default Asterisk AMP & Incoming calls

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:[email protected] 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)


  #6  
Old May 25th 05, 06:29 PM posted to uk.telecom.voip
The Cable Guy
external usenet poster
 
Posts: 55
Default Asterisk AMP & Incoming calls

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:[email protected] 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.....


  #7  
Old May 25th 05, 06:37 PM posted to uk.telecom.voip
Roly
external usenet poster
 
Posts: 25
Default Asterisk AMP & Incoming calls

"Peter Watson" wrote in message
...
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.


  #8  
Old May 25th 05, 06:39 PM posted to uk.telecom.voip
Peter Watson
external usenet poster
 
Posts: 70
Default Asterisk AMP & Incoming calls

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
  #9  
Old May 25th 05, 07:15 PM posted to uk.telecom.voip
Ian
external usenet poster
 
Posts: 289
Default Asterisk AMP & Incoming calls


"Peter Watson" wrote in message
...
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


  #10  
Old May 26th 05, 08:52 PM posted to uk.telecom.voip
Peter Watson
external usenet poster
 
Posts: 70
Default Asterisk AMP & Incoming calls

In article ,
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
 




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